From 551626ec0ad28dc43cae3094c35be7088cc625ab Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 17 Jul 2019 14:30:23 +0200
Subject: ALSA: hda/hdmi - Don't report spurious jack state changes

The HDMI jack handling reports the state change always via
snd_jack_report() whenever hdmi_present_sense() is called, even if the
state itself doesn't change from the previous time.  This is mostly
harmless but still a bit confusing to user-space.

This patch reduces such spurious jack state changes and reports only
when the state really changed.  Also, as a minor optimization, avoid
overwriting the pin ELD data when the state is identical.

Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 32 ++++++++++++++++++--------------
 1 file changed, 18 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bea7b0961080..c380596b2e84 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1421,7 +1421,7 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec,
 /* update per_pin ELD from the given new ELD;
  * setup info frame and notification accordingly
  */
-static void update_eld(struct hda_codec *codec,
+static bool update_eld(struct hda_codec *codec,
 		       struct hdmi_spec_per_pin *per_pin,
 		       struct hdmi_eld *eld)
 {
@@ -1452,18 +1452,22 @@ static void update_eld(struct hda_codec *codec,
 		snd_hdmi_show_eld(codec, &eld->info);
 
 	eld_changed = (pin_eld->eld_valid != eld->eld_valid);
-	if (eld->eld_valid && pin_eld->eld_valid)
+	eld_changed |= (pin_eld->monitor_present != eld->monitor_present);
+	if (!eld_changed && eld->eld_valid && pin_eld->eld_valid)
 		if (pin_eld->eld_size != eld->eld_size ||
 		    memcmp(pin_eld->eld_buffer, eld->eld_buffer,
 			   eld->eld_size) != 0)
 			eld_changed = true;
 
-	pin_eld->monitor_present = eld->monitor_present;
-	pin_eld->eld_valid = eld->eld_valid;
-	pin_eld->eld_size = eld->eld_size;
-	if (eld->eld_valid)
-		memcpy(pin_eld->eld_buffer, eld->eld_buffer, eld->eld_size);
-	pin_eld->info = eld->info;
+	if (eld_changed) {
+		pin_eld->monitor_present = eld->monitor_present;
+		pin_eld->eld_valid = eld->eld_valid;
+		pin_eld->eld_size = eld->eld_size;
+		if (eld->eld_valid)
+			memcpy(pin_eld->eld_buffer, eld->eld_buffer,
+			       eld->eld_size);
+		pin_eld->info = eld->info;
+	}
 
 	/*
 	 * Re-setup pin and infoframe. This is needed e.g. when
@@ -1481,6 +1485,7 @@ static void update_eld(struct hda_codec *codec,
 			       SNDRV_CTL_EVENT_MASK_VALUE |
 			       SNDRV_CTL_EVENT_MASK_INFO,
 			       &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id);
+	return eld_changed;
 }
 
 /* update ELD and jack state via HD-audio verbs */
@@ -1582,6 +1587,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->temp_eld;
 	struct snd_jack *jack = NULL;
+	bool changed;
 	int size;
 
 	mutex_lock(&per_pin->lock);
@@ -1608,15 +1614,13 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
 	 * disconnected event. Jack must be fetched before update_eld()
 	 */
 	jack = pin_idx_to_jack(codec, per_pin);
-	update_eld(codec, per_pin, eld);
+	changed = update_eld(codec, per_pin, eld);
 	if (jack == NULL)
 		jack = pin_idx_to_jack(codec, per_pin);
-	if (jack == NULL)
-		goto unlock;
-	snd_jack_report(jack,
-			(eld->monitor_present && eld->eld_valid) ?
+	if (changed && jack)
+		snd_jack_report(jack,
+				(eld->monitor_present && eld->eld_valid) ?
 				SND_JACK_AVOUT : 0);
- unlock:
 	mutex_unlock(&per_pin->lock);
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From ade49db337a9d44ac5835cfce1ee873549011b27 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 11 Jul 2018 18:05:52 +0200
Subject: ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI

AMD/ATI and Nvidia HDMI codec drivers didn't have the audio component
binding like i915, but it worked only with the traditional HD-audio
unsolicited event for the HDMI hotplug detection and the ELD read-up
thereafter.  This has been a problem in many ways: first of all, it
goes through the hardware event transition (from GPU register write,
HD-audio controller trigger, and finally to HD-audio unsolicited event
handling), which is often unreliable and may miss some opportunities.
Second, each unsol event handling and ELD read-up need the explicit
power up / down when the codec is in the runtime suspend.  Last but
not least, which is the most important, the hotplug wakeup may be
missed when the HD-audio controller is in runtime suspend.  Especially
the last point is a big problem due to the recent change relevant with
vga_switcheroo that forcibly enables the runtime PM for AMD HDMI
controllers.

These issues are solved by introducing the audio component; the
hotplug notification is done by a direct function callback, which is
more accurate and reliable, and it can be processed without the actual
hardware access, i.e. no runtime PM trigger is needed, and the
HD-audio gets the event even if it's in runtime suspend.  The same for
ELD query, as it's read directly from the cached ELD bytes stored in
the DRM driver, hence the whole hardware access can be skipped.

So here it is: this patch implements the audio component binding with
AMD/ATI and Nouveau DRM drivers.  The biggest difference from i915
implementation is that this binding is fully optional and it can be
enabled asynchronously on the fly.  That is, the driver will switch
from the HD-audio unsolicited event to the notify callback once when
the DRM component gets bound.  Similarly, when DRM driver gets
unloaded, the HDMI event handling returns to the legacy mode, too.

Also, another difference from i915 is that the new code registers the
component in the codec driver, while i915 HDMI codec assumes the
component binding was already done in the HD-audio controller driver.
Hence the new code does need to de-register the component binding at
the codec exit, too.

Some other details:
- The match component ops assumes that both VGA and HD-audio
  controller PCI entries belong to the same PCI bus, and only accepts
  such an entry.

- The pin2port audio_ops is implemented with assumption of the fixed
  widget layout.  For AMD, it's starting from 3, with step 2 (3, 5, 7,
  ...), while for Nvidia, it's starting from 4, with step 1 (4, 5, 6,
  ...)

As of this patch, the corresponding component isn't implemented in DRM
side, so this change alone won't give any benefit.  By the following
changes in DRM sides, the mission will be completed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 233 ++++++++++++++++++++++++++++++++++++++++-----
 1 file changed, 211 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index c380596b2e84..2096993eaf28 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -18,6 +18,7 @@
 
 #include <linux/init.h>
 #include <linux/delay.h>
+#include <linux/pci.h>
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <linux/pm_runtime.h>
@@ -119,6 +120,7 @@ struct hdmi_pcm {
 };
 
 struct hdmi_spec {
+	struct hda_codec *codec;
 	int num_cvts;
 	struct snd_array cvts; /* struct hdmi_spec_per_cvt */
 	hda_nid_t cvt_nids[4]; /* only for haswell fix */
@@ -163,9 +165,11 @@ struct hdmi_spec {
 	struct hda_multi_out multiout;
 	struct hda_pcm_stream pcm_playback;
 
-	/* i915/powerwell (Haswell+/Valleyview+) specific */
-	bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */
+	bool use_jack_detect; /* jack detection enabled */
+	bool use_acomp_notifier; /* use eld_notify callback for hotplug */
+	bool acomp_registered; /* audio component registered in this driver */
 	struct drm_audio_component_audio_ops drm_audio_ops;
+	int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */
 
 	struct hdac_chmap chmap;
 	hda_nid_t vendor_nid;
@@ -765,6 +769,10 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
 static void jack_callback(struct hda_codec *codec,
 			  struct hda_jack_callback *jack)
 {
+	/* stop polling when notification is enabled */
+	if (codec_has_acomp(codec))
+		return;
+
 	/* hda_jack don't support DP MST */
 	check_presence_and_report(codec, jack->nid, 0);
 }
@@ -823,6 +831,9 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
 	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
 	int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
 
+	if (codec_has_acomp(codec))
+		return;
+
 	if (!snd_hda_jack_tbl_get_from_tag(codec, tag)) {
 		codec_dbg(codec, "Unexpected HDMI event tag 0x%x\n", tag);
 		return;
@@ -1636,18 +1647,13 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 			snd_hda_power_down_pm(codec);
 			return false;
 		}
-	}
-
-	if (codec_has_acomp(codec)) {
+		ret = hdmi_present_sense_via_verbs(per_pin, repoll);
+		snd_hda_power_down_pm(codec);
+	} else {
 		sync_eld_via_acomp(codec, per_pin);
 		ret = false; /* don't call snd_hda_jack_report_sync() */
-	} else {
-		ret = hdmi_present_sense_via_verbs(per_pin, repoll);
 	}
 
-	if (!codec_has_acomp(codec))
-		snd_hda_power_down_pm(codec);
-
 	return ret;
 }
 
@@ -2252,6 +2258,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx;
 
+	mutex_lock(&spec->pcm_lock);
+	spec->use_jack_detect = !codec->jackpoll_interval;
 	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
 		struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
 		hda_nid_t pin_nid = per_pin->pin_nid;
@@ -2259,11 +2267,15 @@ static int generic_hdmi_init(struct hda_codec *codec)
 
 		snd_hda_set_dev_select(codec, pin_nid, dev_id);
 		hdmi_init_pin(codec, pin_nid);
-		if (!codec_has_acomp(codec))
+		if (codec_has_acomp(codec))
+			continue;
+		if (spec->use_jack_detect)
+			snd_hda_jack_detect_enable(codec, pin_nid);
+		else
 			snd_hda_jack_detect_enable_callback(codec, pin_nid,
-				codec->jackpoll_interval > 0 ?
-				jack_callback : NULL);
+							    jack_callback);
 	}
+	mutex_unlock(&spec->pcm_lock);
 	return 0;
 }
 
@@ -2296,7 +2308,9 @@ static void generic_hdmi_free(struct hda_codec *codec)
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx, pcm_idx;
 
-	if (codec_has_acomp(codec)) {
+	if (spec->acomp_registered) {
+		snd_hdac_acomp_exit(&codec->bus->core);
+	} else if (codec_has_acomp(codec)) {
 		snd_hdac_acomp_register_notifier(&codec->bus->core, NULL);
 		codec->relaxed_resume = 0;
 	}
@@ -2364,6 +2378,7 @@ static int alloc_generic_hdmi(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 
+	spec->codec = codec;
 	spec->ops = generic_standard_hdmi_ops;
 	spec->dev_num = 1;	/* initialize to 1 */
 	mutex_init(&spec->pcm_lock);
@@ -2401,6 +2416,136 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 	return 0;
 }
 
+/*
+ * generic audio component binding
+ */
+
+/* turn on / off the unsol event jack detection dynamically */
+static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid,
+				  bool use_acomp)
+{
+	struct hda_jack_tbl *tbl;
+
+	tbl = snd_hda_jack_tbl_get(codec, nid);
+	if (tbl) {
+		/* clear unsol even if component notifier is used, or re-enable
+		 * if notifier is cleared
+		 */
+		unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag);
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_UNSOLICITED_ENABLE, val);
+	} else {
+		/* if no jack entry was defined beforehand, create a new one
+		 * at need (i.e. only when notifier is cleared)
+		 */
+		if (!use_acomp)
+			snd_hda_jack_detect_enable(codec, nid);
+	}
+}
+
+/* set up / clear component notifier dynamically */
+static void generic_acomp_notifier_set(struct drm_audio_component *acomp,
+				       bool use_acomp)
+{
+	struct hdmi_spec *spec;
+	int i;
+
+	spec = container_of(acomp->audio_ops, struct hdmi_spec, drm_audio_ops);
+	mutex_lock(&spec->pcm_lock);
+	spec->use_acomp_notifier = use_acomp;
+	spec->codec->relaxed_resume = use_acomp;
+	/* reprogram each jack detection logic depending on the notifier */
+	if (spec->use_jack_detect) {
+		for (i = 0; i < spec->num_pins; i++)
+			reprogram_jack_detect(spec->codec,
+					      get_pin(spec, i)->pin_nid,
+					      use_acomp);
+	}
+	mutex_unlock(&spec->pcm_lock);
+}
+
+/* enable / disable the notifier via master bind / unbind */
+static int generic_acomp_master_bind(struct device *dev,
+				     struct drm_audio_component *acomp)
+{
+	generic_acomp_notifier_set(acomp, true);
+	return 0;
+}
+
+static void generic_acomp_master_unbind(struct device *dev,
+					struct drm_audio_component *acomp)
+{
+	generic_acomp_notifier_set(acomp, false);
+}
+
+/* check whether both HD-audio and DRM PCI devices belong to the same bus */
+static int match_bound_vga(struct device *dev, int subtype, void *data)
+{
+	struct hdac_bus *bus = data;
+	struct pci_dev *pci, *master;
+
+	if (!dev_is_pci(dev) || !dev_is_pci(bus->dev))
+		return 0;
+	master = to_pci_dev(bus->dev);
+	pci = to_pci_dev(dev);
+	return master->bus == pci->bus;
+}
+
+/* audio component notifier for AMD/Nvidia HDMI codecs */
+static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id)
+{
+	struct hda_codec *codec = audio_ptr;
+	struct hdmi_spec *spec = codec->spec;
+	hda_nid_t pin_nid = spec->port2pin(codec, port);
+
+	if (!pin_nid)
+		return;
+	if (get_wcaps_type(get_wcaps(codec, pin_nid)) != AC_WID_PIN)
+		return;
+	/* skip notification during system suspend (but not in runtime PM);
+	 * the state will be updated at resume
+	 */
+	if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0)
+		return;
+	/* ditto during suspend/resume process itself */
+	if (snd_hdac_is_in_pm(&codec->core))
+		return;
+
+	check_presence_and_report(codec, pin_nid, dev_id);
+}
+
+/* set up the private drm_audio_ops from the template */
+static void setup_drm_audio_ops(struct hda_codec *codec,
+				const struct drm_audio_component_audio_ops *ops)
+{
+	struct hdmi_spec *spec = codec->spec;
+
+	spec->drm_audio_ops.audio_ptr = codec;
+	/* intel_audio_codec_enable() or intel_audio_codec_disable()
+	 * will call pin_eld_notify with using audio_ptr pointer
+	 * We need make sure audio_ptr is really setup
+	 */
+	wmb();
+	spec->drm_audio_ops.pin2port = ops->pin2port;
+	spec->drm_audio_ops.pin_eld_notify = ops->pin_eld_notify;
+	spec->drm_audio_ops.master_bind = ops->master_bind;
+	spec->drm_audio_ops.master_unbind = ops->master_unbind;
+}
+
+/* initialize the generic HDMI audio component */
+static void generic_acomp_init(struct hda_codec *codec,
+			       const struct drm_audio_component_audio_ops *ops,
+			       int (*port2pin)(struct hda_codec *, int))
+{
+	struct hdmi_spec *spec = codec->spec;
+
+	spec->port2pin = port2pin;
+	setup_drm_audio_ops(codec, ops);
+	if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
+				 match_bound_vga, 0))
+		spec->acomp_registered = true;
+}
+
 /*
  * Intel codec parsers and helpers
  */
@@ -2569,20 +2714,19 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe)
 	check_presence_and_report(codec, pin_nid, dev_id);
 }
 
+static const struct drm_audio_component_audio_ops intel_audio_ops = {
+	.pin2port = intel_pin2port,
+	.pin_eld_notify = intel_pin_eld_notify,
+};
+
 /* register i915 component pin_eld_notify callback */
 static void register_i915_notifier(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
 
 	spec->use_acomp_notifier = true;
-	spec->drm_audio_ops.audio_ptr = codec;
-	/* intel_audio_codec_enable() or intel_audio_codec_disable()
-	 * will call pin_eld_notify with using audio_ptr pointer
-	 * We need make sure audio_ptr is really setup
-	 */
-	wmb();
-	spec->drm_audio_ops.pin2port = intel_pin2port;
-	spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify;
+	spec->port2pin = intel_port2pin;
+	setup_drm_audio_ops(codec, &intel_audio_ops);
 	snd_hdac_acomp_register_notifier(&codec->bus->core,
 					&spec->drm_audio_ops);
 	/* no need for forcible resume for jack check thanks to notifier */
@@ -2980,6 +3124,7 @@ static int patch_simple_hdmi(struct hda_codec *codec,
 	if (!spec)
 		return -ENOMEM;
 
+	spec->codec = codec;
 	codec->spec = spec;
 	hdmi_array_init(spec, 1);
 
@@ -3284,6 +3429,26 @@ static int nvhdmi_chmap_validate(struct hdac_chmap *chmap,
 	return 0;
 }
 
+/* map from pin NID to port; port is 0-based */
+/* for Nvidia: assume widget NID starting from 4, with step 1 (4, 5, 6, ...) */
+static int nvhdmi_pin2port(void *audio_ptr, int pin_nid)
+{
+	return pin_nid - 4;
+}
+
+/* reverse-map from port to pin NID: see above */
+static int nvhdmi_port2pin(struct hda_codec *codec, int port)
+{
+	return port + 4;
+}
+
+static const struct drm_audio_component_audio_ops nvhdmi_audio_ops = {
+	.pin2port = nvhdmi_pin2port,
+	.pin_eld_notify = generic_acomp_pin_eld_notify,
+	.master_bind = generic_acomp_master_bind,
+	.master_unbind = generic_acomp_master_unbind,
+};
+
 static int patch_nvhdmi(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
@@ -3300,6 +3465,8 @@ static int patch_nvhdmi(struct hda_codec *codec)
 		nvhdmi_chmap_cea_alloc_validate_get_type;
 	spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
 
+	generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin);
+
 	return 0;
 }
 
@@ -3787,6 +3954,26 @@ static int atihdmi_init(struct hda_codec *codec)
 	return 0;
 }
 
+/* map from pin NID to port; port is 0-based */
+/* for AMD: assume widget NID starting from 3, with step 2 (3, 5, 7, ...) */
+static int atihdmi_pin2port(void *audio_ptr, int pin_nid)
+{
+	return pin_nid / 2 - 1;
+}
+
+/* reverse-map from port to pin NID: see above */
+static int atihdmi_port2pin(struct hda_codec *codec, int port)
+{
+	return port * 2 + 3;
+}
+
+static const struct drm_audio_component_audio_ops atihdmi_audio_ops = {
+	.pin2port = atihdmi_pin2port,
+	.pin_eld_notify = generic_acomp_pin_eld_notify,
+	.master_bind = generic_acomp_master_bind,
+	.master_unbind = generic_acomp_master_unbind,
+};
+
 static int patch_atihdmi(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
@@ -3835,6 +4022,8 @@ static int patch_atihdmi(struct hda_codec *codec)
 	 */
 	codec->link_down_at_suspend = 1;
 
+	generic_acomp_init(codec, &atihdmi_audio_ops, atihdmi_port2pin);
+
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From caa8422d01e983782548648e125fd617cadcec3f Mon Sep 17 00:00:00 2001
From: Chris Wilson <chris@chris-wilson.co.uk>
Date: Sat, 20 Jul 2019 12:33:37 +0100
Subject: ALSA: hda: Flush interrupts on disabling

I was looking at

<4> [241.835158] general protection fault: 0000 [#1] PREEMPT SMP PTI
<4> [241.835181] CPU: 1 PID: 214 Comm: kworker/1:3 Tainted: G     U            5.2.0-CI-CI_DRM_6509+ #1
<4> [241.835199] Hardware name: Dell Inc.                 OptiPlex 745                 /0GW726, BIOS 2.3.1  05/21/2007
<4> [241.835234] Workqueue: events snd_hdac_bus_process_unsol_events [snd_hda_core]
<4> [241.835256] RIP: 0010:input_handle_event+0x16d/0x5e0
<4> [241.835270] Code: 48 8b 93 58 01 00 00 8b 52 08 89 50 04 8b 83 f8 06 00 00 48 8b 93 00 07 00 00 8d 70 01 48 8d 04 c2 83 e1 08 89 b3 f8 06 00 00 <66> 89 28 66 44 89 60 02 44 89 68 04 8b 93 f8 06 00 00 0f 84 fd fe
<4> [241.835304] RSP: 0018:ffffc9000019fda0 EFLAGS: 00010046
<4> [241.835317] RAX: 6b6b6b6ec6c6c6c3 RBX: ffff8880290fefc8 RCX: 0000000000000000
<4> [241.835332] RDX: 000000006b6b6b6b RSI: 000000006b6b6b6c RDI: 0000000000000046
<4> [241.835347] RBP: 0000000000000005 R08: 0000000000000000 R09: 0000000000000001
<4> [241.835362] R10: ffffc9000019faa0 R11: 0000000000000000 R12: 0000000000000004
<4> [241.835377] R13: 0000000000000000 R14: ffff8880290ff1d0 R15: 0000000000000293
<4> [241.835392] FS:  0000000000000000(0000) GS:ffff88803de80000(0000) knlGS:0000000000000000
<4> [241.835409] CS:  0010 DS: 0000 ES: 0000 CR0: 0000000080050033
<4> [241.835422] CR2: 00007ffe9a99e9b7 CR3: 000000002f588000 CR4: 00000000000006e0
<4> [241.835436] Call Trace:
<4> [241.835449]  input_event+0x45/0x70
<4> [241.835464]  snd_jack_report+0xdc/0x100
<4> [241.835490]  snd_hda_jack_report_sync+0x83/0xc0 [snd_hda_codec]
<4> [241.835512]  snd_hdac_bus_process_unsol_events+0x5a/0x70 [snd_hda_core]
<4> [241.835530]  process_one_work+0x245/0x610

which has the hallmarks of a worker queued from interrupt after it was
supposedly cancelled (note the POISON_FREE), and I could not see where
the interrupt would be flushed on shutdown so added the likely suspects.

Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=111174
Signed-off-by: Chris Wilson <chris@chris-wilson.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_controller.c | 2 ++
 sound/pci/hda/hda_intel.c   | 2 +-
 2 files changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 3b0110545070..196bbc85699e 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -447,6 +447,8 @@ static void azx_int_disable(struct hdac_bus *bus)
 	list_for_each_entry(azx_dev, &bus->stream_list, list)
 		snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0);
 
+	synchronize_irq(bus->irq);
+
 	/* disable SIE for all streams */
 	snd_hdac_chip_writeb(bus, INTCTL, 0);
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cb8b0945547c..fdde80d95966 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1301,9 +1301,9 @@ static int azx_free(struct azx *chip)
 	}
 
 	if (bus->chip_init) {
+		azx_stop_chip(chip);
 		azx_clear_irq_pending(chip);
 		azx_stop_all_streams(chip);
-		azx_stop_chip(chip);
 	}
 
 	if (bus->irq >= 0)
-- 
cgit v1.2.3-59-g8ed1b


From ef9bec27485fefb6b93168fea73fda0dc9638046 Mon Sep 17 00:00:00 2001
From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Date: Wed, 26 Jun 2019 00:04:50 -0700
Subject: ASoC: SOF: Intel: hda: Make hdac_device device-managed

snd_hdac_ext_bus_device_exit() has been recently modified
to no longer free the hdac device. SOF allocates memory for
hdac_device and hda_hda_priv with kzalloc. Make them
device-managed instead so that they will be freed when the
SOF driver is unloaded.

Because of the above change, hda_codec is device-managed and
it will be freed when the ASoC device is removed. Freeing
the codec in snd_hda_codec_dev_release() leads to kernel
panic while unloading and reloading the ASoC driver. So,
avoid freeing the hda_codec for ASoC driver. This is done in
the same patch to avoid bisect failure.

Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20190626070450.7229-1-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/pci/hda/hda_codec.c       | 8 +++++++-
 sound/soc/sof/intel/hda-codec.c | 6 ++----
 2 files changed, 9 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e30e86ca6b72..133200d31170 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -846,7 +846,13 @@ static void snd_hda_codec_dev_release(struct device *dev)
 	snd_hda_sysfs_clear(codec);
 	kfree(codec->modelname);
 	kfree(codec->wcaps);
-	kfree(codec);
+
+	/*
+	 * In the case of ASoC HD-audio, hda_codec is device managed.
+	 * It will be freed when the ASoC device is removed.
+	 */
+	if (codec->core.type == HDA_DEV_LEGACY)
+		kfree(codec);
 }
 
 #define DEV_NAME_LEN 31
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index b8b37f082309..0d8437b080bf 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -62,8 +62,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address)
 		address, resp);
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
-	/* snd_hdac_ext_bus_device_exit will use kfree to free hdev */
-	hda_priv = kzalloc(sizeof(*hda_priv), GFP_KERNEL);
+	hda_priv = devm_kzalloc(sdev->dev, sizeof(*hda_priv), GFP_KERNEL);
 	if (!hda_priv)
 		return -ENOMEM;
 
@@ -82,8 +81,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address)
 
 	return 0;
 #else
-	/* snd_hdac_ext_bus_device_exit will use kfree to free hdev */
-	hdev = kzalloc(sizeof(*hdev), GFP_KERNEL);
+	hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL);
 	if (!hdev)
 		return -ENOMEM;
 
-- 
cgit v1.2.3-59-g8ed1b


From f7408a3d5b5fd10571a653d1a81ce9167c62727f Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Tue, 9 Jul 2019 22:04:25 -0700
Subject: ASoC: wcd9335: Fix misuse of GENMASK macro

Arguments are supposed to be ordered high then low.

Signed-off-by: Joe Perches <joe@perches.com>
Link: https://lore.kernel.org/r/92e31a9f321fe731d428ec3ec9d4654ea8a16d1b.1562734889.git.joe@perches.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd-clsh-v2.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wcd-clsh-v2.c b/sound/soc/codecs/wcd-clsh-v2.c
index c397d713f01a..cc5a9c9b918b 100644
--- a/sound/soc/codecs/wcd-clsh-v2.c
+++ b/sound/soc/codecs/wcd-clsh-v2.c
@@ -65,7 +65,7 @@ struct wcd_clsh_ctrl {
 #define WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY			0
 #define WCD9XXX_RX_BIAS_FLYB_BUFF			WCD9335_REG(0x6, 0xC7)
 #define WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK		GENMASK(7, 4)
-#define WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK		GENMASK(0, 3)
+#define WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK		GENMASK(3, 0)
 #define WCD9XXX_HPH_L_EN				WCD9335_REG(0x6, 0xD3)
 #define WCD9XXX_HPH_CONST_SEL_L_MASK			GENMASK(7, 3)
 #define WCD9XXX_HPH_CONST_SEL_BYPASS			0
-- 
cgit v1.2.3-59-g8ed1b


From eaba5585944e6d692828bca701c9362f60172b8a Mon Sep 17 00:00:00 2001
From: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Date: Wed, 10 Jul 2019 13:51:19 +0300
Subject: ASoC: codecs: ad193x: Use regmap_multi_reg_write() when initializing

Using regmap_multi_reg_write() when we set the default values for our
registers makes the code smaller and easier to read.

Suggested-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190710105119.22987-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ad193x.c | 19 +++++--------------
 1 file changed, 5 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 80dab5df9633..fb04c9379b71 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -413,15 +413,10 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = {
 	.ops = &ad193x_dai_ops,
 };
 
-struct ad193x_reg_default {
-	unsigned int reg;
-	unsigned int val;
-};
-
 /* codec register values to set after reset */
 static void ad193x_reg_default_init(struct ad193x_priv *ad193x)
 {
-	const struct ad193x_reg_default reg_init[] = {
+	const struct reg_sequence reg_init[] = {
 		{  0, 0x99 },	/* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */
 		{  1, 0x04 },	/* PLL_CLK_CTRL1: no on-chip Vref */
 		{  2, 0x40 },	/* DAC_CTRL0: TDM mode */
@@ -437,21 +432,17 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x)
 		{ 12, 0x00 },	/* DAC_L4_VOL: no attenuation */
 		{ 13, 0x00 },	/* DAC_R4_VOL: no attenuation */
 	};
-	const struct ad193x_reg_default reg_adc_init[] = {
+	const struct reg_sequence reg_adc_init[] = {
 		{ 14, 0x03 },	/* ADC_CTRL0: high-pass filter enable */
 		{ 15, 0x43 },	/* ADC_CTRL1: sata delay=1, adc aux mode */
 		{ 16, 0x00 },	/* ADC_CTRL2: reset */
 	};
-	int i;
 
-	for (i = 0; i < ARRAY_SIZE(reg_init); i++)
-		regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val);
+	regmap_multi_reg_write(ad193x->regmap, reg_init, ARRAY_SIZE(reg_init));
 
 	if (ad193x_has_adc(ad193x)) {
-		for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) {
-			regmap_write(ad193x->regmap, reg_adc_init[i].reg,
-				     reg_adc_init[i].val);
-		}
+		regmap_multi_reg_write(ad193x->regmap, reg_adc_init,
+				       ARRAY_SIZE(reg_adc_init));
 	}
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 4dc057a786dc23fd1ed70e6bcab691c0f26daa1e Mon Sep 17 00:00:00 2001
From: Curtis Malainey <cujomalainey@chromium.org>
Date: Wed, 10 Jul 2019 17:28:38 -0700
Subject: ASoC: rt5677: remove superfluous set

We define this variable with the same value, no need to set it twice

Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20190711002838.35363-1-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index c779dc3474f9..deaed5132dc9 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -5098,7 +5098,6 @@ static irqreturn_t rt5677_irq(int unused, void *data)
 			goto exit;
 		}
 
-		irq_fired = false;
 		for (i = 0; i < RT5677_IRQ_NUM; i++) {
 			if (reg_irq & rt5677_irq_descs[i].status_mask) {
 				irq_fired = true;
-- 
cgit v1.2.3-59-g8ed1b


From 5be6155b50bbf7083b4bfa219e4ce6d1491f42f0 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Thu, 11 Jul 2019 18:49:45 +0800
Subject: ASoC: fsl_esai: Wrap some operations to be functions

Extract the operation to be functions, to improve the
readability.

In this patch, fsl_esai_hw_init, fsl_esai_register_restore,
fsl_esai_trigger_start and fsl_esai_trigger_stop are
extracted.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/804d7e75ae7e06a913479912b578b3538ca7cd3f.1562842206.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_esai.c | 188 +++++++++++++++++++++++++++++------------------
 1 file changed, 117 insertions(+), 71 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 10d2210c91ef..ab460d6d7432 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -35,6 +35,7 @@
  * @fifo_depth: depth of tx/rx FIFO
  * @slot_width: width of each DAI slot
  * @slots: number of slots
+ * @channels: channel num for tx or rx
  * @hck_rate: clock rate of desired HCKx clock
  * @sck_rate: clock rate of desired SCKx clock
  * @hck_dir: the direction of HCKx pads
@@ -57,6 +58,7 @@ struct fsl_esai {
 	u32 slots;
 	u32 tx_mask;
 	u32 rx_mask;
+	u32 channels[2];
 	u32 hck_rate[2];
 	u32 sck_rate[2];
 	bool hck_dir[2];
@@ -543,64 +545,132 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int fsl_esai_hw_init(struct fsl_esai *esai_priv)
+{
+	struct platform_device *pdev = esai_priv->pdev;
+	int ret;
+
+	/* Reset ESAI unit */
+	ret = regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR,
+				 ESAI_ECR_ESAIEN_MASK | ESAI_ECR_ERST_MASK,
+				 ESAI_ECR_ESAIEN | ESAI_ECR_ERST);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to reset ESAI: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * We need to enable ESAI so as to access some of its registers.
+	 * Otherwise, we would fail to dump regmap from user space.
+	 */
+	ret = regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR,
+				 ESAI_ECR_ESAIEN_MASK | ESAI_ECR_ERST_MASK,
+				 ESAI_ECR_ESAIEN);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to enable ESAI: %d\n", ret);
+		return ret;
+	}
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
+			   ESAI_PRRC_PDC_MASK, 0);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
+			   ESAI_PCRC_PC_MASK, 0);
+
+	return 0;
+}
+
+static int fsl_esai_register_restore(struct fsl_esai *esai_priv)
+{
+	int ret;
+
+	/* FIFO reset for safety */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR,
+			   ESAI_xFCR_xFR, ESAI_xFCR_xFR);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR,
+			   ESAI_xFCR_xFR, ESAI_xFCR_xFR);
+
+	regcache_mark_dirty(esai_priv->regmap);
+	ret = regcache_sync(esai_priv->regmap);
+	if (ret)
+		return ret;
+
+	/* FIFO reset done */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0);
+
+	return 0;
+}
+
+static void fsl_esai_trigger_start(struct fsl_esai *esai_priv, bool tx)
+{
+	u8 i, channels = esai_priv->channels[tx];
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
+	u32 mask;
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+			   ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN);
+
+	/* Write initial words reqiured by ESAI as normal procedure */
+	for (i = 0; tx && i < channels; i++)
+		regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0);
+
+	/*
+	 * When set the TE/RE in the end of enablement flow, there
+	 * will be channel swap issue for multi data line case.
+	 * In order to workaround this issue, we switch the bit
+	 * enablement sequence to below sequence
+	 * 1) clear the xSMB & xSMA: which is done in probe and
+	 *                           stop state.
+	 * 2) set TE/RE
+	 * 3) set xSMB
+	 * 4) set xSMA:  xSMA is the last one in this flow, which
+	 *               will trigger esai to start.
+	 */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+			   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
+			   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
+	mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask;
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
+			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask));
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
+			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask));
+}
+
+static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx)
+{
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+			   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
+			   ESAI_xSMA_xS_MASK, 0);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
+			   ESAI_xSMB_xS_MASK, 0);
+
+	/* Disable and reset FIFO */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+			   ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+			   ESAI_xFCR_xFR, 0);
+}
+
 static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 			    struct snd_soc_dai *dai)
 {
 	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	u8 i, channels = substream->runtime->channels;
-	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
-	u32 mask;
+
+	esai_priv->channels[tx] = substream->runtime->channels;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
-				   ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN);
-
-		/* Write initial words reqiured by ESAI as normal procedure */
-		for (i = 0; tx && i < channels; i++)
-			regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0);
-
-		/*
-		 * When set the TE/RE in the end of enablement flow, there
-		 * will be channel swap issue for multi data line case.
-		 * In order to workaround this issue, we switch the bit
-		 * enablement sequence to below sequence
-		 * 1) clear the xSMB & xSMA: which is done in probe and
-		 *                           stop state.
-		 * 2) set TE/RE
-		 * 3) set xSMB
-		 * 4) set xSMA:  xSMA is the last one in this flow, which
-		 *               will trigger esai to start.
-		 */
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
-				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
-				   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
-		mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask;
-
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
-				   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask));
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
-				   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask));
-
+		fsl_esai_trigger_start(esai_priv, tx);
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
-				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0);
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
-				   ESAI_xSMA_xS_MASK, 0);
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx),
-				   ESAI_xSMB_xS_MASK, 0);
-
-		/* Disable and reset FIFO */
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
-				   ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR);
-		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
-				   ESAI_xFCR_xFR, 0);
+		fsl_esai_trigger_stop(esai_priv, tx);
 		break;
 	default:
 		return -EINVAL;
@@ -866,22 +936,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
 
 	dev_set_drvdata(&pdev->dev, esai_priv);
 
-	/* Reset ESAI unit */
-	ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST);
-	if (ret) {
-		dev_err(&pdev->dev, "failed to reset ESAI: %d\n", ret);
-		return ret;
-	}
-
-	/*
-	 * We need to enable ESAI so as to access some of its registers.
-	 * Otherwise, we would fail to dump regmap from user space.
-	 */
-	ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN);
-	if (ret) {
-		dev_err(&pdev->dev, "failed to enable ESAI: %d\n", ret);
+	ret = fsl_esai_hw_init(esai_priv);
+	if (ret)
 		return ret;
-	}
 
 	esai_priv->tx_mask = 0xFFFFFFFF;
 	esai_priv->rx_mask = 0xFFFFFFFF;
@@ -955,20 +1012,10 @@ static int fsl_esai_runtime_resume(struct device *dev)
 
 	regcache_cache_only(esai->regmap, false);
 
-	/* FIFO reset for safety */
-	regmap_update_bits(esai->regmap, REG_ESAI_TFCR,
-			   ESAI_xFCR_xFR, ESAI_xFCR_xFR);
-	regmap_update_bits(esai->regmap, REG_ESAI_RFCR,
-			   ESAI_xFCR_xFR, ESAI_xFCR_xFR);
-
-	ret = regcache_sync(esai->regmap);
+	ret = fsl_esai_register_restore(esai);
 	if (ret)
 		goto err_regcache_sync;
 
-	/* FIFO reset done */
-	regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0);
-	regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0);
-
 	return 0;
 
 err_regcache_sync:
@@ -991,7 +1038,6 @@ static int fsl_esai_runtime_suspend(struct device *dev)
 	struct fsl_esai *esai = dev_get_drvdata(dev);
 
 	regcache_cache_only(esai->regmap, true);
-	regcache_mark_dirty(esai->regmap);
 
 	if (!IS_ERR(esai->fsysclk))
 		clk_disable_unprepare(esai->fsysclk);
-- 
cgit v1.2.3-59-g8ed1b


From 1d3dd532883be6167da5df6117efd6d4e8790456 Mon Sep 17 00:00:00 2001
From: Hariprasad Kelam <hariprasad.kelam@gmail.com>
Date: Tue, 9 Jul 2019 23:55:43 +0530
Subject: sound: soc: codecs: mt6358: change return type of
 mt6358_codec_init_reg

As mt6358_codec_init_reg function always returns 0 , change return type
from int to void.

fixes below issue reported by coccicheck
sound/soc/codecs/mt6358.c:2260:5-8: Unneeded variable: "ret". Return "0"
on line 2289

Signed-off-by: Hariprasad Kelam <hariprasad.kelam@gmail.com>
Acked-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190709182543.GA6611@hari-Inspiron-1545
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/mt6358.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c
index 50b3fc5457ea..c17250aab2d0 100644
--- a/sound/soc/codecs/mt6358.c
+++ b/sound/soc/codecs/mt6358.c
@@ -2255,10 +2255,8 @@ static struct snd_soc_dai_driver mt6358_dai_driver[] = {
 	},
 };
 
-static int mt6358_codec_init_reg(struct mt6358_priv *priv)
+static void mt6358_codec_init_reg(struct mt6358_priv *priv)
 {
-	int ret = 0;
-
 	/* Disable HeadphoneL/HeadphoneR short circuit protection */
 	regmap_update_bits(priv->regmap, MT6358_AUDDEC_ANA_CON0,
 			   RG_AUDHPLSCDISABLE_VAUDP15_MASK_SFT,
@@ -2285,8 +2283,6 @@ static int mt6358_codec_init_reg(struct mt6358_priv *priv)
 	/* set gpio */
 	playback_gpio_reset(priv);
 	capture_gpio_reset(priv);
-
-	return ret;
 }
 
 static int mt6358_codec_probe(struct snd_soc_component *cmpnt)
-- 
cgit v1.2.3-59-g8ed1b


From e1fb714bddf452ef0ecf67c420957438977f107d Mon Sep 17 00:00:00 2001
From: Hariprasad Kelam <hariprasad.kelam@gmail.com>
Date: Wed, 10 Jul 2019 00:12:37 +0530
Subject: ASoC: bcm: cygnus-pcm: Unneeded variable: "ret".

This patch fixes below issues reported by coccicheck

sound/soc/bcm/cygnus-pcm.c:642:5-8: Unneeded variable: "ret". Return "0"
on line 650
sound/soc/bcm/cygnus-pcm.c:671:5-8: Unneeded variable: "ret". Return "0"
on line 696

We cannot change return type of these functions as they are callback
functions of snd_pcm_ops

Signed-off-by: Hariprasad Kelam <hariprasad.kelam@gmail.com>
Link: https://lore.kernel.org/r/20190709184236.GA7873@hari-Inspiron-1545
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/bcm/cygnus-pcm.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c
index 123ecf5479d7..8966b02844dc 100644
--- a/sound/soc/bcm/cygnus-pcm.c
+++ b/sound/soc/bcm/cygnus-pcm.c
@@ -639,7 +639,6 @@ static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct cygnus_aio_port *aio;
-	int ret = 0;
 
 	aio = cygnus_dai_get_dma_data(substream);
 	dev_dbg(rtd->cpu_dai->dev, "%s  port %d\n", __func__, aio->portnum);
@@ -647,7 +646,7 @@ static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream,
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = params_buffer_bytes(params);
 
-	return ret;
+	return 0;
 }
 
 static int cygnus_pcm_hw_free(struct snd_pcm_substream *substream)
@@ -668,7 +667,6 @@ static int cygnus_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct cygnus_aio_port *aio;
 	unsigned long bufsize, periodsize;
-	int ret = 0;
 	bool is_play;
 	u32 start;
 	struct ringbuf_regs *p_rbuf = NULL;
@@ -693,7 +691,7 @@ static int cygnus_pcm_prepare(struct snd_pcm_substream *substream)
 	ringbuf_set_initial(aio->cygaud->audio, p_rbuf, is_play, start,
 				periodsize, bufsize);
 
-	return ret;
+	return 0;
 }
 
 static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_pcm_substream *substream)
-- 
cgit v1.2.3-59-g8ed1b


From af62a3c8a872631e63833e2a99a8f43f81532fae Mon Sep 17 00:00:00 2001
From: Hariprasad Kelam <hariprasad.kelam@gmail.com>
Date: Wed, 10 Jul 2019 07:46:27 +0530
Subject: ASoC: wcd9335: add irqflag IRQF_ONESHOT flag

Add IRQF_ONESHOT to ensure "Interrupt is not reenabled after the hardirq
handler finished".

fixes below issue reported by coccicheck

sound/soc/codecs/wcd9335.c:4068:8-33: ERROR: Threaded IRQ with no
primary handler requested without IRQF_ONESHOT

Signed-off-by: Hariprasad Kelam <hariprasad.kelam@gmail.com>
Link: https://lore.kernel.org/r/20190710021627.GA13396@hari-Inspiron-1545
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd9335.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index 1bbbe421b999..956602788d0e 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -4062,7 +4062,8 @@ static int wcd9335_setup_irqs(struct wcd9335_codec *wcd)
 
 		ret = devm_request_threaded_irq(wcd->dev, irq, NULL,
 						wcd9335_irqs[i].handler,
-						IRQF_TRIGGER_RISING,
+						IRQF_TRIGGER_RISING |
+						IRQF_ONESHOT,
 						wcd9335_irqs[i].name, wcd);
 		if (ret) {
 			dev_err(wcd->dev, "Failed to request %s\n",
-- 
cgit v1.2.3-59-g8ed1b


From a962a809e5e45715081fa247726a562b8dbd082a Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Tue, 16 Jul 2019 11:24:17 +0800
Subject: ASoC: mediatek: mt8183: make headset codec optional

Make headset codec optional because some variant machines may not
have an audio jack.

Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190716032417.19015-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 887c932229d0..4c816c86844b 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -352,8 +352,6 @@ static struct snd_soc_card mt8183_mt6358_ts3a227_max98357_card = {
 	.owner = THIS_MODULE,
 	.dai_link = mt8183_mt6358_ts3a227_max98357_dai_links,
 	.num_links = ARRAY_SIZE(mt8183_mt6358_ts3a227_max98357_dai_links),
-	.aux_dev = &mt8183_mt6358_ts3a227_max98357_headset_dev,
-	.num_aux_devs = 1,
 };
 
 static int
@@ -404,10 +402,9 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev)
 	mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node =
 		of_parse_phandle(pdev->dev.of_node,
 				 "mediatek,headset-codec", 0);
-	if (!mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) {
-		dev_err(&pdev->dev,
-			"Property 'mediatek,headset-codec' missing/invalid\n");
-		return -EINVAL;
+	if (mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) {
+		card->aux_dev = &mt8183_mt6358_ts3a227_max98357_headset_dev;
+		card->num_aux_devs = 1;
 	}
 
 	default_pins =
-- 
cgit v1.2.3-59-g8ed1b


From 89c9679f699d88986ce552738dc7c5c500c8fc67 Mon Sep 17 00:00:00 2001
From: Lucas Stach <l.stach@pengutronix.de>
Date: Wed, 17 Jul 2019 12:56:33 +0200
Subject: ASoC: fsl_sai: add of_match data

New revisions of the SAI IP block have even more differences that need
be taken into account by the driver. To avoid sprinking compatible
checks all over the driver move the current differences into of_match_data.

Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Tested-by: Angus Ainslie <angus@akkea.ca>
Reviewed-by: Angus Ainslie <angus@akkea.ca>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20190717105635.18514-2-l.stach@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 22 ++++++++++++++--------
 sound/soc/fsl/fsl_sai.h |  6 +++++-
 2 files changed, 19 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index d58cc3ae90d8..ed0432e7327a 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -9,6 +9,7 @@
 #include <linux/dmaengine.h>
 #include <linux/module.h>
 #include <linux/of_address.h>
+#include <linux/of_device.h>
 #include <linux/pm_runtime.h>
 #include <linux/regmap.h>
 #include <linux/slab.h>
@@ -788,10 +789,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	sai->pdev = pdev;
-
-	if (of_device_is_compatible(np, "fsl,imx6sx-sai") ||
-	    of_device_is_compatible(np, "fsl,imx6ul-sai"))
-		sai->sai_on_imx = true;
+	sai->soc_data = of_device_get_match_data(&pdev->dev);
 
 	sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
 
@@ -900,7 +898,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	if (sai->sai_on_imx)
+	if (sai->soc_data->use_imx_pcm)
 		return imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE);
 	else
 		return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
@@ -913,10 +911,18 @@ static int fsl_sai_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct fsl_sai_soc_data fsl_sai_vf610_data = {
+	.use_imx_pcm = false,
+};
+
+static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
+	.use_imx_pcm = true,
+};
+
 static const struct of_device_id fsl_sai_ids[] = {
-	{ .compatible = "fsl,vf610-sai", },
-	{ .compatible = "fsl,imx6sx-sai", },
-	{ .compatible = "fsl,imx6ul-sai", },
+	{ .compatible = "fsl,vf610-sai", .data = &fsl_sai_vf610_data },
+	{ .compatible = "fsl,imx6sx-sai", .data = &fsl_sai_imx6sx_data },
+	{ .compatible = "fsl,imx6ul-sai", .data = &fsl_sai_imx6sx_data },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(of, fsl_sai_ids);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 24cb156bf995..83e2bfe05b1b 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -126,6 +126,10 @@
 #define FSL_SAI_MAXBURST_TX 6
 #define FSL_SAI_MAXBURST_RX 6
 
+struct fsl_sai_soc_data {
+	bool use_imx_pcm;
+};
+
 struct fsl_sai {
 	struct platform_device *pdev;
 	struct regmap *regmap;
@@ -135,7 +139,6 @@ struct fsl_sai {
 	bool is_slave_mode;
 	bool is_lsb_first;
 	bool is_dsp_mode;
-	bool sai_on_imx;
 	bool synchronous[2];
 
 	unsigned int mclk_id[2];
@@ -143,6 +146,7 @@ struct fsl_sai {
 	unsigned int slots;
 	unsigned int slot_width;
 
+	const struct fsl_sai_soc_data *soc_data;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 };
-- 
cgit v1.2.3-59-g8ed1b


From bd517707d85f19a7339ea8b882fcbf0fd9976bd6 Mon Sep 17 00:00:00 2001
From: Lucas Stach <l.stach@pengutronix.de>
Date: Wed, 17 Jul 2019 12:56:34 +0200
Subject: ASoC: fsl_sai: derive TX FIFO watermark from FIFO depth

The DMA request schould be triggered as soon as the FIFO has space
for another burst. As different versions of the SAI block have
different FIFO sizes, the watrmark level needs to be derived from
version specific data.

Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Tested-by: Angus Ainslie <angus@akkea.ca>
Reviewed-by: Angus Ainslie <angus@akkea.ca>
Link: https://lore.kernel.org/r/20190717105635.18514-3-l.stach@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 4 +++-
 sound/soc/fsl/fsl_sai.h | 1 +
 2 files changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ed0432e7327a..1d1a447163e3 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -640,7 +640,7 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
 
 	regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
-			   FSL_SAI_MAXBURST_TX * 2);
+			   sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
 			   FSL_SAI_MAXBURST_RX - 1);
 
@@ -913,10 +913,12 @@ static int fsl_sai_remove(struct platform_device *pdev)
 
 static const struct fsl_sai_soc_data fsl_sai_vf610_data = {
 	.use_imx_pcm = false,
+	.fifo_depth = 32,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
 	.use_imx_pcm = true,
+	.fifo_depth = 32,
 };
 
 static const struct of_device_id fsl_sai_ids[] = {
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 83e2bfe05b1b..7c1ef671da28 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -128,6 +128,7 @@
 
 struct fsl_sai_soc_data {
 	bool use_imx_pcm;
+	unsigned int fifo_depth;
 };
 
 struct fsl_sai {
-- 
cgit v1.2.3-59-g8ed1b


From 6d19d8a3cec74a9680947ecb6abdeda38583110e Mon Sep 17 00:00:00 2001
From: Lucas Stach <l.stach@pengutronix.de>
Date: Wed, 17 Jul 2019 12:51:56 +0200
Subject: ASoC: fsl_sai: mark regmap as fast_io

The regmap is only ever used to access MMIO registers, so it's fair
to say that register access is fast.

Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Link: https://lore.kernel.org/r/20190717105156.15721-1-l.stach@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 1d1a447163e3..6d3c6c8d50ce 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -763,6 +763,7 @@ static const struct regmap_config fsl_sai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
+	.fast_io = true,
 
 	.max_register = FSL_SAI_RMR,
 	.reg_defaults = fsl_sai_reg_defaults,
-- 
cgit v1.2.3-59-g8ed1b


From 96cd3b97bf32557ba722c2e96095cb6361b5dc42 Mon Sep 17 00:00:00 2001
From: fengchunguo <chunguo.feng@amlogic.com>
Date: Mon, 15 Jul 2019 17:55:56 +0800
Subject: ASoC: max98383: fix i2c probe failure

Added reset_gpio configuration for i2c probe successfully.
If not,i2c address can't be found rightly.

Error information:
max98373 3-0031: Failed to read: 0x21FF

Fixed:
[3.761299@3] max98373 3-0031: MAX98373 revisionID: 0x43
[3.828911@3] asoc-aml-card auge_sound: max98373-aif1 <-> TDM-B mapping ok

Signed-off-by: fengchunguo <chunguo.feng@amlogic.com>
Link: https://lore.kernel.org/r/20190715095556.1614-1-chunguo.feng@amlogic.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++----
 sound/soc/codecs/max98373.h |  1 +
 2 files changed, 31 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 528695cd6a1c..9a1eb7222357 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -12,6 +12,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <linux/gpio.h>
+#include <linux/of.h>
 #include <linux/of_gpio.h>
 #include <sound/tlv.h>
 #include "max98373.h"
@@ -895,6 +896,17 @@ static void max98373_slot_config(struct i2c_client *i2c,
 	else
 		max98373->i_slot = 1;
 
+	max98373->reset_gpio = of_get_named_gpio(dev->of_node,
+						"maxim,reset-gpio", 0);
+	if (!gpio_is_valid(max98373->reset_gpio)) {
+		dev_err(dev, "Looking up %s property in node %s failed %d\n",
+			"maxim,reset-gpio", dev->of_node->full_name,
+			max98373->reset_gpio);
+	} else {
+		dev_dbg(dev, "maxim,reset-gpio=%d",
+			max98373->reset_gpio);
+	}
+
 	if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value))
 		max98373->spkfb_slot = value & 0xF;
 	else
@@ -923,7 +935,6 @@ static int max98373_i2c_probe(struct i2c_client *i2c,
 	else
 		max98373->interleave_mode = false;
 
-
 	/* regmap initialization */
 	max98373->regmap
 		= devm_regmap_init_i2c(i2c, &max98373_regmap);
@@ -934,6 +945,24 @@ static int max98373_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
+	/* voltage/current slot & gpio configuration */
+	max98373_slot_config(i2c, max98373);
+
+	/* Power on device */
+	if (gpio_is_valid(max98373->reset_gpio)) {
+		ret = gpio_request(max98373->reset_gpio, "MAX98373_RESET");
+		if (ret) {
+			dev_err(&i2c->dev, "%s: Failed to request gpio %d\n",
+				__func__, max98373->reset_gpio);
+			gpio_free(max98373->reset_gpio);
+			return -EINVAL;
+		}
+		gpio_direction_output(max98373->reset_gpio, 0);
+		msleep(50);
+		gpio_direction_output(max98373->reset_gpio, 1);
+		msleep(20);
+	}
+
 	/* Check Revision ID */
 	ret = regmap_read(max98373->regmap,
 		MAX98373_R21FF_REV_ID, &reg);
@@ -944,9 +973,6 @@ static int max98373_i2c_probe(struct i2c_client *i2c,
 	}
 	dev_info(&i2c->dev, "MAX98373 revisionID: 0x%02X\n", reg);
 
-	/* voltage/current slot configuration */
-	max98373_slot_config(i2c, max98373);
-
 	/* codec registeration */
 	ret = devm_snd_soc_register_component(&i2c->dev, &soc_codec_dev_max98373,
 		max98373_dai, ARRAY_SIZE(max98373_dai));
diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h
index f6a37aa02f26..533d2053f608 100644
--- a/sound/soc/codecs/max98373.h
+++ b/sound/soc/codecs/max98373.h
@@ -203,6 +203,7 @@
 
 struct max98373_priv {
 	struct regmap *regmap;
+	int reset_gpio;
 	unsigned int v_slot;
 	unsigned int i_slot;
 	unsigned int spkfb_slot;
-- 
cgit v1.2.3-59-g8ed1b


From b7e814deae33eb30f8f8c6528e8e69b107978d88 Mon Sep 17 00:00:00 2001
From: Lucas Stach <l.stach@pengutronix.de>
Date: Fri, 19 Jul 2019 16:36:37 +0200
Subject: ASoC: tlv320aic31xx: suppress error message for EPROBE_DEFER

Both the supplies and reset GPIO might need a probe deferral for the
resource to be available. Don't print a error message in that case, as
it is a normal operating condition.

Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Acked-by: Andrew F. Davis <afd@ti.com>
Link: https://lore.kernel.org/r/20190719143637.2018-1-l.stach@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic31xx.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 9b37e98da0db..26a4f6cd3288 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1553,7 +1553,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 	aic31xx->gpio_reset = devm_gpiod_get_optional(aic31xx->dev, "reset",
 						      GPIOD_OUT_LOW);
 	if (IS_ERR(aic31xx->gpio_reset)) {
-		dev_err(aic31xx->dev, "not able to acquire gpio\n");
+		if (PTR_ERR(aic31xx->gpio_reset) != -EPROBE_DEFER)
+			dev_err(aic31xx->dev, "not able to acquire gpio\n");
 		return PTR_ERR(aic31xx->gpio_reset);
 	}
 
@@ -1564,7 +1565,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 				      ARRAY_SIZE(aic31xx->supplies),
 				      aic31xx->supplies);
 	if (ret) {
-		dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
+		if (ret != -EPROBE_DEFER)
+			dev_err(aic31xx->dev,
+				"Failed to request supplies: %d\n", ret);
 		return ret;
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From 980c10cda954770640d9d7cd7b793efa12cae0d9 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Fri, 19 Jul 2019 14:32:35 +0800
Subject: ASoC: rt1308: simplify the EFUSE read function

The rt1308_efuse function contains many redundant settings.
We remove the redundant settings and the function still works.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20190719063235.18757-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1308.c | 22 ----------------------
 1 file changed, 22 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
index d673506c7c39..19d255e01827 100755
--- a/sound/soc/codecs/rt1308.c
+++ b/sound/soc/codecs/rt1308.c
@@ -808,33 +808,11 @@ static void rt1308_efuse(struct rt1308_priv *rt1308)
 {
 	regmap_write(rt1308->regmap, RT1308_RESET, 0);
 
-	regmap_write(rt1308->regmap, RT1308_POWER, 0xff371600);
-	regmap_write(rt1308->regmap, RT1308_CLK_1, 0x52100000);
-	regmap_write(rt1308->regmap, RT1308_I2C_I2S_SDW_SET, 0x01014005);
-	regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x227f5501);
-	regmap_write(rt1308->regmap, RT1308_PADS_1, 0x50150505);
-	regmap_write(rt1308->regmap, RT1308_VREF, 0x18100000);
-	regmap_write(rt1308->regmap, RT1308_IV_SENSE, 0x87010000);
-	regmap_write(rt1308->regmap, RT1308_DUMMY_REG, 0x00000200);
-	regmap_write(rt1308->regmap, RT1308_SIL_DET, 0x61c30000);
-	regmap_write(rt1308->regmap, RT1308_CLK_DET, 0x03700000);
-	regmap_write(rt1308->regmap, RT1308_SINE_TONE_GEN_1, 0x50022f00);
 	regmap_write(rt1308->regmap, RT1308_POWER_STATUS, 0x01800000);
-	regmap_write(rt1308->regmap, RT1308_DC_CAL_2, 0x00ffff00);
-	regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x607e5501);
-
-	regmap_write(rt1308->regmap, RT1308_CLK_2, 0x0060e000);
-	regmap_write(rt1308->regmap, RT1308_EFUSE_1, 0x04fe0f00);
 	msleep(100);
 	regmap_write(rt1308->regmap, RT1308_EFUSE_1, 0x44fe0f00);
 	msleep(20);
 	regmap_write(rt1308->regmap, RT1308_PVDD_OFFSET_CTL, 0x10000000);
-
-	regmap_write(rt1308->regmap, RT1308_POWER_STATUS, 0x00800000);
-	regmap_write(rt1308->regmap, RT1308_POWER, 0x0);
-	regmap_write(rt1308->regmap, RT1308_CLK_1, 0x52000000);
-	regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x227f5501);
-	regmap_write(rt1308->regmap, RT1308_SINE_TONE_GEN_1, 0x10022f00);
 }
 
 static int rt1308_i2c_probe(struct i2c_client *i2c,
-- 
cgit v1.2.3-59-g8ed1b


From 730e051c78b05f0ce87df06e0fc1ea002d722fe8 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Fri, 19 Jul 2019 14:32:49 +0800
Subject: ASoC: rt1308: add silence detection and manual PDB control

We enable the silence detection function in initial settings.
PDB control changes to manual mode, hence the driver could
fully control the AMP output on/off.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20190719063249.18806-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1308.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
index 19d255e01827..cf3d72f99705 100755
--- a/sound/soc/codecs/rt1308.c
+++ b/sound/soc/codecs/rt1308.c
@@ -40,10 +40,10 @@ static const struct reg_sequence init_list[] = {
 	{ RT1308_VREF,			0x18100000 },
 	{ RT1308_IV_SENSE,		0x87010000 },
 	{ RT1308_DUMMY_REG,		0x00000200 },
-	{ RT1308_SIL_DET,		0x61c30000 },
+	{ RT1308_SIL_DET,		0xe1c30000 },
 	{ RT1308_DC_CAL_2,		0x00ffff00 },
 	{ RT1308_CLK_DET,		0x01000000 },
-	{ RT1308_POWER_STATUS,		0x00800000 },
+	{ RT1308_POWER_STATUS,		0x08800000 },
 	{ RT1308_DAC_SET,		0xafaf0700 },
 
 };
@@ -308,12 +308,13 @@ static int rt1308_classd_event(struct snd_soc_dapm_widget *w,
 	case SND_SOC_DAPM_POST_PMU:
 		msleep(30);
 		snd_soc_component_update_bits(component, RT1308_POWER_STATUS,
-			RT1308_POW_PDB_REG_BIT, RT1308_POW_PDB_REG_BIT);
+			RT1308_POW_PDB_REG_BIT | RT1308_POW_PDB_MN_BIT,
+			RT1308_POW_PDB_REG_BIT | RT1308_POW_PDB_MN_BIT);
 		msleep(40);
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
 		snd_soc_component_update_bits(component, RT1308_POWER_STATUS,
-			RT1308_POW_PDB_REG_BIT, 0);
+			RT1308_POW_PDB_REG_BIT | RT1308_POW_PDB_MN_BIT, 0);
 		usleep_range(150000, 200000);
 		break;
 
-- 
cgit v1.2.3-59-g8ed1b


From 03b0deb6ba5f47d346fbb0783018c0d415203142 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Mon, 22 Jul 2019 10:35:16 +0100
Subject: ASoC: cs47l35: Use define for DSPCLK enable bit

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190722093516.12906-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs47l35.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c
index e3585c1dab3d..d396a8545d51 100644
--- a/sound/soc/codecs/cs47l35.c
+++ b/sound/soc/codecs/cs47l35.c
@@ -524,7 +524,7 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT,
 		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK,
 		    MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, 6,
+SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, MADERA_DSP_CLK_ENA_SHIFT,
 		    0, NULL, 0),
 
 SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0),
-- 
cgit v1.2.3-59-g8ed1b


From 4aefb7387198740318c5e7bf7211701f4e0d849e Mon Sep 17 00:00:00 2001
From: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Date: Fri, 19 Jul 2019 20:05:58 +0200
Subject: ASoC: rockchip-max98090: Remove MICBIAS as supply of input pin IN34

Commit ec0d23b295b9 ("ASoC: rockchip-max98090: Fix the Headset Mic
route.") moved the MICBIAS widget to supply Headset Mic but forget to
remove the MICBIAS widget to supply IN34 which is not really needed, so
remove that path so we have:

   IN34 <==== Headset MIC <==== MICBIAS

Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Link: https://lore.kernel.org/r/20190719180558.11459-1-enric.balletbo@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_max98090.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c
index c5fc24675a33..d1c907631c2d 100644
--- a/sound/soc/rockchip/rockchip_max98090.c
+++ b/sound/soc/rockchip/rockchip_max98090.c
@@ -45,7 +45,6 @@ static const struct snd_soc_dapm_widget rk_dapm_widgets[] = {
 
 static const struct snd_soc_dapm_route rk_audio_map[] = {
 	{"IN34", NULL, "Headset Mic"},
-	{"IN34", NULL, "MICBIAS"},
 	{"Headset Mic", NULL, "MICBIAS"},
 	{"DMICL", NULL, "Int Mic"},
 	{"Headphone", NULL, "HPL"},
-- 
cgit v1.2.3-59-g8ed1b


From b1f373a11d25fc9a5f7679c9b85799fe09b0dc4a Mon Sep 17 00:00:00 2001
From: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Date: Fri, 19 Jul 2019 10:05:31 +0000
Subject: ASoC: sgtl5000: Improve VAG power and mute control

VAG power control is improved to fit the manual [1]. This patch fixes as
minimum one bug: if customer muxes Headphone to Line-In right after boot,
the VAG power remains off that leads to poor sound quality from line-in.

I.e. after boot:
  - Connect sound source to Line-In jack;
  - Connect headphone to HP jack;
  - Run following commands:
  $ amixer set 'Headphone' 80%
  $ amixer set 'Headphone Mux' LINE_IN

Change VAG power on/off control according to the following algorithm:
  - turn VAG power ON on the 1st incoming event.
  - keep it ON if there is any active VAG consumer (ADC/DAC/HP/Line-In).
  - turn VAG power OFF when there is the latest consumer's pre-down event
    come.
  - always delay after VAG power OFF to avoid pop.
  - delay after VAG power ON if the initiative consumer is Line-In, this
    prevents pop during line-in muxing.

According to the data sheet [1], to avoid any pops/clicks,
the outputs should be muted during input/output
routing changes.

[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf

Cc: stable@vger.kernel.org
Fixes: 9b34e6cc3bc2 ("ASoC: Add Freescale SGTL5000 codec support")
Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190719100524.23300-3-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 224 ++++++++++++++++++++++++++++++++++++++------
 1 file changed, 194 insertions(+), 30 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index a6a4748c97f9..34cc85e49003 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -31,6 +31,13 @@
 #define SGTL5000_DAP_REG_OFFSET	0x0100
 #define SGTL5000_MAX_REG_OFFSET	0x013A
 
+/* Delay for the VAG ramp up */
+#define SGTL5000_VAG_POWERUP_DELAY 500 /* ms */
+/* Delay for the VAG ramp down */
+#define SGTL5000_VAG_POWERDOWN_DELAY 500 /* ms */
+
+#define SGTL5000_OUTPUTS_MUTE (SGTL5000_HP_MUTE | SGTL5000_LINE_OUT_MUTE)
+
 /* default value of sgtl5000 registers */
 static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_CHIP_DIG_POWER,		0x0000 },
@@ -123,6 +130,13 @@ enum  {
 	I2S_SCLK_STRENGTH_HIGH,
 };
 
+enum {
+	HP_POWER_EVENT,
+	DAC_POWER_EVENT,
+	ADC_POWER_EVENT,
+	LAST_POWER_EVENT = ADC_POWER_EVENT
+};
+
 /* sgtl5000 private structure in codec */
 struct sgtl5000_priv {
 	int sysclk;	/* sysclk rate */
@@ -137,8 +151,109 @@ struct sgtl5000_priv {
 	u8 micbias_voltage;
 	u8 lrclk_strength;
 	u8 sclk_strength;
+	u16 mute_state[LAST_POWER_EVENT + 1];
 };
 
+static inline int hp_sel_input(struct snd_soc_component *component)
+{
+	return (snd_soc_component_read32(component, SGTL5000_CHIP_ANA_CTRL) &
+		SGTL5000_HP_SEL_MASK) >> SGTL5000_HP_SEL_SHIFT;
+}
+
+static inline u16 mute_output(struct snd_soc_component *component,
+			      u16 mute_mask)
+{
+	u16 mute_reg = snd_soc_component_read32(component,
+					      SGTL5000_CHIP_ANA_CTRL);
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+			    mute_mask, mute_mask);
+	return mute_reg;
+}
+
+static inline void restore_output(struct snd_soc_component *component,
+				  u16 mute_mask, u16 mute_reg)
+{
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+		mute_mask, mute_reg);
+}
+
+static void vag_power_on(struct snd_soc_component *component, u32 source)
+{
+	if (snd_soc_component_read32(component, SGTL5000_CHIP_ANA_POWER) &
+	    SGTL5000_VAG_POWERUP)
+		return;
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+			    SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+
+	/* When VAG powering on to get local loop from Line-In, the sleep
+	 * is required to avoid loud pop.
+	 */
+	if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN &&
+	    source == HP_POWER_EVENT)
+		msleep(SGTL5000_VAG_POWERUP_DELAY);
+}
+
+static int vag_power_consumers(struct snd_soc_component *component,
+			       u16 ana_pwr_reg, u32 source)
+{
+	int consumers = 0;
+
+	/* count dac/adc consumers unconditional */
+	if (ana_pwr_reg & SGTL5000_DAC_POWERUP)
+		consumers++;
+	if (ana_pwr_reg & SGTL5000_ADC_POWERUP)
+		consumers++;
+
+	/*
+	 * If the event comes from HP and Line-In is selected,
+	 * current action is 'DAC to be powered down'.
+	 * As HP_POWERUP is not set when HP muxed to line-in,
+	 * we need to keep VAG power ON.
+	 */
+	if (source == HP_POWER_EVENT) {
+		if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN)
+			consumers++;
+	} else {
+		if (ana_pwr_reg & SGTL5000_HP_POWERUP)
+			consumers++;
+	}
+
+	return consumers;
+}
+
+static void vag_power_off(struct snd_soc_component *component, u32 source)
+{
+	u16 ana_pwr = snd_soc_component_read32(component,
+					     SGTL5000_CHIP_ANA_POWER);
+
+	if (!(ana_pwr & SGTL5000_VAG_POWERUP))
+		return;
+
+	/*
+	 * This function calls when any of VAG power consumers is disappearing.
+	 * Thus, if there is more than one consumer at the moment, as minimum
+	 * one consumer will definitely stay after the end of the current
+	 * event.
+	 * Don't clear VAG_POWERUP if 2 or more consumers of VAG present:
+	 * - LINE_IN (for HP events) / HP (for DAC/ADC events)
+	 * - DAC
+	 * - ADC
+	 * (the current consumer is disappearing right now)
+	 */
+	if (vag_power_consumers(component, ana_pwr, source) >= 2)
+		return;
+
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
+		SGTL5000_VAG_POWERUP, 0);
+	/* In power down case, we need wait 400-1000 ms
+	 * when VAG fully ramped down.
+	 * As longer we wait, as smaller pop we've got.
+	 */
+	msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+}
+
 /*
  * mic_bias power on/off share the same register bits with
  * output impedance of mic bias, when power on mic bias, we
@@ -170,36 +285,46 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-/*
- * As manual described, ADC/DAC only works when VAG powerup,
- * So enabled VAG before ADC/DAC up.
- * In power down case, we need wait 400ms when vag fully ramped down.
- */
-static int power_vag_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
+static int vag_and_mute_control(struct snd_soc_component *component,
+				 int event, int event_source)
 {
-	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-	const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+	static const u16 mute_mask[] = {
+		/*
+		 * Mask for HP_POWER_EVENT.
+		 * Muxing Headphones have to be wrapped with mute/unmute
+		 * headphones only.
+		 */
+		SGTL5000_HP_MUTE,
+		/*
+		 * Masks for DAC_POWER_EVENT/ADC_POWER_EVENT.
+		 * Muxing DAC or ADC block have to wrapped with mute/unmute
+		 * both headphones and line-out.
+		 */
+		SGTL5000_OUTPUTS_MUTE,
+		SGTL5000_OUTPUTS_MUTE
+	};
+
+	struct sgtl5000_priv *sgtl5000 =
+		snd_soc_component_get_drvdata(component);
 
 	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		sgtl5000->mute_state[event_source] =
+			mute_output(component, mute_mask[event_source]);
+		break;
 	case SND_SOC_DAPM_POST_PMU:
-		snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
-			SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
-		msleep(400);
+		vag_power_on(component, event_source);
+		restore_output(component, mute_mask[event_source],
+			       sgtl5000->mute_state[event_source]);
 		break;
-
 	case SND_SOC_DAPM_PRE_PMD:
-		/*
-		 * Don't clear VAG_POWERUP, when both DAC and ADC are
-		 * operational to prevent inadvertently starving the
-		 * other one of them.
-		 */
-		if ((snd_soc_component_read32(component, SGTL5000_CHIP_ANA_POWER) &
-				mask) != mask) {
-			snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER,
-				SGTL5000_VAG_POWERUP, 0);
-			msleep(400);
-		}
+		sgtl5000->mute_state[event_source] =
+			mute_output(component, mute_mask[event_source]);
+		vag_power_off(component, event_source);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		restore_output(component, mute_mask[event_source],
+			       sgtl5000->mute_state[event_source]);
 		break;
 	default:
 		break;
@@ -208,6 +333,41 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+/*
+ * Mute Headphone when power it up/down.
+ * Control VAG power on HP power path.
+ */
+static int headphone_pga_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, HP_POWER_EVENT);
+}
+
+/* As manual describes, ADC/DAC powering up/down requires
+ * to mute outputs to avoid pops.
+ * Control VAG power on ADC/DAC power path.
+ */
+static int adc_updown_depop(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, ADC_POWER_EVENT);
+}
+
+static int dac_updown_depop(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	return vag_and_mute_control(component, event, DAC_POWER_EVENT);
+}
+
 /* input sources for ADC */
 static const char *adc_mux_text[] = {
 	"MIC_IN", "LINE_IN"
@@ -280,7 +440,10 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 			    mic_bias_event,
 			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
-	SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
+			   headphone_pga_event,
+			   SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
 	SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
@@ -301,11 +464,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
 				0, SGTL5000_CHIP_DIG_POWER,
 				1, 0),
 
-	SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
-	SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
-
-	SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event),
-	SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event),
+	SND_SOC_DAPM_ADC_E("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0,
+			   adc_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
+	SND_SOC_DAPM_DAC_E("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0,
+			   dac_updown_depop, SND_SOC_DAPM_PRE_POST_PMU |
+			   SND_SOC_DAPM_PRE_POST_PMD),
 };
 
 /* routes for sgtl5000 */
-- 
cgit v1.2.3-59-g8ed1b


From cadee10c983145c16d6c75af88f6a943b22ebeb2 Mon Sep 17 00:00:00 2001
From: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Date: Fri, 19 Jul 2019 10:05:33 +0000
Subject: ASoC: sgtl5000: Fix definition of VAG Ramp Control

SGTL5000_SMALL_POP is a bit mask, not a value. Usage of
correct definition makes device probing code more clear.

Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Reviewed-by: Igor Opaniuk <igor.opaniuk@toradex.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20190719100524.23300-4-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 2 +-
 sound/soc/codecs/sgtl5000.h | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 34cc85e49003..3f28e7862b5b 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1460,7 +1460,7 @@ static int sgtl5000_probe(struct snd_soc_component *component)
 
 	/* enable small pop, introduce 400ms delay in turning off */
 	snd_soc_component_update_bits(component, SGTL5000_CHIP_REF_CTRL,
-				SGTL5000_SMALL_POP, 1);
+				SGTL5000_SMALL_POP, SGTL5000_SMALL_POP);
 
 	/* disable short cut detector */
 	snd_soc_component_write(component, SGTL5000_CHIP_SHORT_CTRL, 0);
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index 18cae08bbd3a..a4bf4bca95bf 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -273,7 +273,7 @@
 #define SGTL5000_BIAS_CTRL_MASK			0x000e
 #define SGTL5000_BIAS_CTRL_SHIFT		1
 #define SGTL5000_BIAS_CTRL_WIDTH		3
-#define SGTL5000_SMALL_POP			1
+#define SGTL5000_SMALL_POP			0x0001
 
 /*
  * SGTL5000_CHIP_MIC_CTRL
-- 
cgit v1.2.3-59-g8ed1b


From 694b14554d75f2a1ae111202e71860d58b434a21 Mon Sep 17 00:00:00 2001
From: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Date: Fri, 19 Jul 2019 10:05:34 +0000
Subject: ASoC: sgtl5000: add ADC mute control

This control mute/unmute the ADC input of SGTL5000
using its CHIP_ANA_CTRL register.

Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Reviewed-by: Igor Opaniuk <igor.opaniuk@toradex.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20190719100524.23300-5-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 3f28e7862b5b..b65232521ea8 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -720,6 +720,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 			SGTL5000_CHIP_ANA_ADC_CTRL,
 			8, 1, 0, capture_6db_attenuate),
 	SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
+	SOC_SINGLE("Capture Switch", SGTL5000_CHIP_ANA_CTRL, 0, 1, 1),
 
 	SOC_DOUBLE_TLV("Headphone Playback Volume",
 			SGTL5000_CHIP_ANA_HP_CTRL,
-- 
cgit v1.2.3-59-g8ed1b


From 631bc8f0134ae9620d86a96b8c5f9445d91a2dca Mon Sep 17 00:00:00 2001
From: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Date: Fri, 19 Jul 2019 10:05:35 +0000
Subject: ASoC: sgtl5000: Fix of unmute outputs on probe

To enable "zero cross detect" for ADC/HP, change
HP_ZCD_EN/ADC_ZCD_EN bits only instead of writing the whole
CHIP_ANA_CTRL register.

Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Reviewed-by: Igor Opaniuk <igor.opaniuk@toradex.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20190719100524.23300-6-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index b65232521ea8..23f4ae2f0723 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1453,6 +1453,7 @@ static int sgtl5000_probe(struct snd_soc_component *component)
 	int ret;
 	u16 reg;
 	struct sgtl5000_priv *sgtl5000 = snd_soc_component_get_drvdata(component);
+	unsigned int zcd_mask = SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN;
 
 	/* power up sgtl5000 */
 	ret = sgtl5000_set_power_regs(component);
@@ -1480,9 +1481,8 @@ static int sgtl5000_probe(struct snd_soc_component *component)
 	       0x1f);
 	snd_soc_component_write(component, SGTL5000_CHIP_PAD_STRENGTH, reg);
 
-	snd_soc_component_write(component, SGTL5000_CHIP_ANA_CTRL,
-			SGTL5000_HP_ZCD_EN |
-			SGTL5000_ADC_ZCD_EN);
+	snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL,
+		zcd_mask, zcd_mask);
 
 	snd_soc_component_update_bits(component, SGTL5000_CHIP_MIC_CTRL,
 			SGTL5000_BIAS_R_MASK,
-- 
cgit v1.2.3-59-g8ed1b


From b6319b061ba279577fd7030a9848fbd6a17151e3 Mon Sep 17 00:00:00 2001
From: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Date: Fri, 19 Jul 2019 10:05:37 +0000
Subject: ASoC: sgtl5000: Fix charge pump source assignment

If VDDA != VDDIO and any of them is greater than 3.1V, charge pump
source can be assigned automatically [1].

[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf

Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Reviewed-by: Igor Opaniuk <igor.opaniuk@toradex.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20190719100524.23300-7-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 15 ++++++++++-----
 1 file changed, 10 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 23f4ae2f0723..aa1f9637d895 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1338,12 +1338,17 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component)
 					SGTL5000_INT_OSC_EN);
 		/* Enable VDDC charge pump */
 		ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
-	} else if (vddio >= 3100 && vdda >= 3100) {
+	} else {
 		ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP;
-		/* VDDC use VDDIO rail */
-		lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
-		lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
-			    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+		/*
+		 * if vddio == vdda the source of charge pump should be
+		 * assigned manually to VDDIO
+		 */
+		if (vddio == vdda) {
+			lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
+			lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
+				    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+		}
 	}
 
 	snd_soc_component_write(component, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
-- 
cgit v1.2.3-59-g8ed1b


From 588f2e2caf6795ca29d50a45ea1e1438274e75e0 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:51 +0900
Subject: ALSA: firewire-lib: obsolete ctx_data.tx.first_dbc with
 CIP_UNALIGHED_DBC flag

Recent firmware for Fireworks board module have a quirk to start
transmission of CIP with non-zero value for its data block counter.
In current implementation of ALSA firewire stack, the quirk is handled
by 'struct amdtp_stream.ctx_data.tx.first_dbc' with value 0x02. However,
the value comes from reverse engineering. It's better to handle this
quirk without the explicit value.

In a process to parse CIP header, the quirk of data block counter
affects decision of sequence index in sequence-multiplexed data channel;
i.e. MIDI conformant data channel. In Fireworks, the index is decided
by the number of data blocks from top of the same CIP, thus the value
of data block counter is useless.

This commit adds CIP_UNALIGHED_DBC flag and obsoletes the explicit
value for this quirk.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-am824.c                | 8 ++++++--
 sound/firewire/amdtp-stream.c               | 3 +--
 sound/firewire/amdtp-stream.h               | 5 +++--
 sound/firewire/fireworks/fireworks_stream.c | 2 +-
 4 files changed, 11 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index fd5d6b8ac557..99c567ded7a3 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -315,12 +315,16 @@ static void read_midi_messages(struct amdtp_stream *s,
 			       __be32 *buffer, unsigned int frames)
 {
 	struct amdtp_am824 *p = s->protocol;
-	unsigned int f, port;
 	int len;
 	u8 *b;
+	int f;
 
 	for (f = 0; f < frames; f++) {
-		port = (8 - s->ctx_data.tx.first_dbc + s->data_block_counter + f) % 8;
+		unsigned int port = f;
+
+		if (!(s->flags & CIP_UNALIGHED_DBC))
+			port += s->data_block_counter;
+		port %= 8;
 		b = (u8 *)&buffer[p->midi_position];
 
 		len = b[0] - 0x80;
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 4d71d74707cf..fc1e8e5b9429 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -584,8 +584,7 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
 	    s->data_block_counter != UINT_MAX)
 		*dbc = s->data_block_counter;
 
-	if (((s->flags & CIP_SKIP_DBC_ZERO_CHECK) &&
-	     *dbc == s->ctx_data.tx.first_dbc) ||
+	if ((*dbc == 0x00 && (s->flags & CIP_SKIP_DBC_ZERO_CHECK)) ||
 	    s->data_block_counter == UINT_MAX) {
 		lost = false;
 	} else if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 3942894c11ac..5d611122312b 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -33,6 +33,8 @@
  * @CIP_HEADER_WITHOUT_EOH: Only for in-stream. CIP Header doesn't include
  *	valid EOH.
  * @CIP_NO_HEADERS: a lack of headers in packets
+ * @CIP_UNALIGHED_DBC: Only for in-stream. The value of dbc is not alighed to
+ *	the value of current SYT_INTERVAL; e.g. initial value is not zero.
  */
 enum cip_flags {
 	CIP_NONBLOCKING		= 0x00,
@@ -45,6 +47,7 @@ enum cip_flags {
 	CIP_JUMBO_PAYLOAD	= 0x40,
 	CIP_HEADER_WITHOUT_EOH	= 0x80,
 	CIP_NO_HEADER		= 0x100,
+	CIP_UNALIGHED_DBC	= 0x200,
 };
 
 /**
@@ -119,8 +122,6 @@ struct amdtp_stream {
 			// Fixed interval of dbc between previos/current
 			// packets.
 			unsigned int dbc_interval;
-			// Indicate the value of dbc field in a first packet.
-			unsigned int first_dbc;
 		} tx;
 		struct {
 			// To calculate CIP data blocks and tstamp.
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index e659a0b89ba5..385fc9686365 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -146,7 +146,7 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw)
 	    (efw->firmware_version == 0x5070000 ||
 	     efw->firmware_version == 0x5070300 ||
 	     efw->firmware_version == 0x5080000))
-		efw->tx_stream.ctx_data.tx.first_dbc = 0x02;
+		efw->tx_stream.flags |= CIP_UNALIGHED_DBC;
 	/* AudioFire9 always reports wrong dbs. */
 	if (efw->is_af9)
 		efw->tx_stream.flags |= CIP_WRONG_DBS;
-- 
cgit v1.2.3-59-g8ed1b


From cdefaa44d535a9d555ac4f1925d59ae4d0ef98e8 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:52 +0900
Subject: ALSA: firewire-lib: pass data block count as an argument to
 tracepoints event

This is a preparation for future commit that 'struct
amdtp_stream.data_block_count' does not represent the value of
data block count for current data block.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream-trace.h | 6 +++---
 sound/firewire/amdtp-stream.c       | 4 ++--
 2 files changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h
index 4adbbf789cbe..16c7f6605511 100644
--- a/sound/firewire/amdtp-stream-trace.h
+++ b/sound/firewire/amdtp-stream-trace.h
@@ -14,8 +14,8 @@
 #include <linux/tracepoint.h>
 
 TRACE_EVENT(amdtp_packet,
-	TP_PROTO(const struct amdtp_stream *s, u32 cycles, const __be32 *cip_header, unsigned int payload_length, unsigned int data_blocks, unsigned int index),
-	TP_ARGS(s, cycles, cip_header, payload_length, data_blocks, index),
+	TP_PROTO(const struct amdtp_stream *s, u32 cycles, const __be32 *cip_header, unsigned int payload_length, unsigned int data_blocks, unsigned int data_block_counter, unsigned int index),
+	TP_ARGS(s, cycles, cip_header, payload_length, data_blocks, data_block_counter, index),
 	TP_STRUCT__entry(
 		__field(unsigned int, second)
 		__field(unsigned int, cycle)
@@ -47,7 +47,7 @@ TRACE_EVENT(amdtp_packet,
 		}
 		__entry->payload_quadlets = payload_length / sizeof(__be32);
 		__entry->data_blocks = data_blocks;
-		__entry->data_block_counter = s->data_block_counter,
+		__entry->data_block_counter = data_block_counter,
 		__entry->packet_index = s->packet_index;
 		__entry->irq = !!in_interrupt();
 		__entry->index = index;
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index fc1e8e5b9429..fe1512905a43 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -511,7 +511,7 @@ static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle,
 	}
 
 	trace_amdtp_packet(s, cycle, cip_header, payload_length, data_blocks,
-			   index);
+			   s->data_block_counter, index);
 
 	if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
 		s->data_block_counter =
@@ -653,7 +653,7 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle,
 	s->data_block_counter = dbc;
 
 	trace_amdtp_packet(s, cycle, cip_header, *payload_length, *data_blocks,
-			   index);
+			   s->data_block_counter, index);
 
 	return err;
 }
-- 
cgit v1.2.3-59-g8ed1b


From ab75481202d2925a017e4cb5407b0bb49c356de8 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:53 +0900
Subject: ALSA: firewire-lib: pass data block counter to data block processing
 layer

This is a preparation for future commit that 'struct
amdtp_stream.data_block_count' does not represent the value of
data block count for current data block.

However, data block count is required for calculation of sequence index
in sequence-multiplied data channel. Some of data block processing layer
require it; e.g. for AM824 data block.

This commit passes data block count to the processing layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-am824.c         | 24 +++++++++++++-----------
 sound/firewire/amdtp-stream.c        |  4 ++--
 sound/firewire/amdtp-stream.h        |  1 +
 sound/firewire/digi00x/amdtp-dot.c   | 16 +++++++---------
 sound/firewire/fireface/amdtp-ff.c   | 10 ++++------
 sound/firewire/motu/amdtp-motu.c     |  8 ++++----
 sound/firewire/tascam/amdtp-tascam.c | 10 ++++------
 7 files changed, 35 insertions(+), 38 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 99c567ded7a3..03a7e0533131 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -284,7 +284,7 @@ static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
 }
 
 static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
-				unsigned int frames)
+			unsigned int frames, unsigned int data_block_counter)
 {
 	struct amdtp_am824 *p = s->protocol;
 	unsigned int f, port;
@@ -293,7 +293,7 @@ static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
 	for (f = 0; f < frames; f++) {
 		b = (u8 *)&buffer[p->midi_position];
 
-		port = (s->data_block_counter + f) % 8;
+		port = (data_block_counter + f) % 8;
 		if (f < MAX_MIDI_RX_BLOCKS &&
 		    midi_ratelimit_per_packet(s, port) &&
 		    p->midi[port] != NULL &&
@@ -311,8 +311,8 @@ static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
 	}
 }
 
-static void read_midi_messages(struct amdtp_stream *s,
-			       __be32 *buffer, unsigned int frames)
+static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+			unsigned int frames, unsigned int data_block_counter)
 {
 	struct amdtp_am824 *p = s->protocol;
 	int len;
@@ -323,7 +323,7 @@ static void read_midi_messages(struct amdtp_stream *s,
 		unsigned int port = f;
 
 		if (!(s->flags & CIP_UNALIGHED_DBC))
-			port += s->data_block_counter;
+			port += data_block_counter;
 		port %= 8;
 		b = (u8 *)&buffer[p->midi_position];
 
@@ -335,8 +335,9 @@ static void read_midi_messages(struct amdtp_stream *s,
 	}
 }
 
-static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
-					   unsigned int data_blocks, unsigned int *syt)
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct amdtp_am824 *p = s->protocol;
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
@@ -351,13 +352,14 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffe
 	}
 
 	if (p->midi_ports)
-		write_midi_messages(s, buffer, data_blocks);
+		write_midi_messages(s, buffer, data_blocks, data_block_counter);
 
 	return pcm_frames;
 }
 
-static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
-					   unsigned int data_blocks, unsigned int *syt)
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct amdtp_am824 *p = s->protocol;
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
@@ -371,7 +373,7 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffe
 	}
 
 	if (p->midi_ports)
-		read_midi_messages(s, buffer, data_blocks);
+		read_midi_messages(s, buffer, data_blocks, data_block_counter);
 
 	return pcm_frames;
 }
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index fe1512905a43..40de9fc9d751 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -722,7 +722,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		data_blocks = calculate_data_blocks(s, syt);
 		buffer = s->buffer.packets[s->packet_index].buffer;
 		pcm_frames = s->process_data_blocks(s, buffer, data_blocks,
-						    &syt);
+						s->data_block_counter, &syt);
 
 		build_it_pkt_header(s, cycle, &template.params, data_blocks,
 				    syt, i);
@@ -776,7 +776,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		if (err >= 0) {
 			buffer = s->buffer.packets[s->packet_index].buffer;
 			pcm_frames = s->process_data_blocks(s, buffer,
-							    data_blocks, &syt);
+				data_blocks, s->data_block_counter, &syt);
 
 			if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
 				s->data_block_counter += data_blocks;
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 5d611122312b..762ac3c7e902 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -99,6 +99,7 @@ typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
 						struct amdtp_stream *s,
 						__be32 *buffer,
 						unsigned int data_blocks,
+						unsigned int data_block_counter,
 						unsigned int *syt);
 struct amdtp_stream {
 	struct fw_unit *unit;
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
index 45ff73d16074..5c12973c2652 100644
--- a/sound/firewire/digi00x/amdtp-dot.c
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -234,7 +234,7 @@ static inline void midi_use_bytes(struct amdtp_stream *s,
 }
 
 static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
-				unsigned int data_blocks)
+		unsigned int data_blocks, unsigned int data_block_counter)
 {
 	struct amdtp_dot *p = s->protocol;
 	unsigned int f, port;
@@ -242,7 +242,7 @@ static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
 	u8 *b;
 
 	for (f = 0; f < data_blocks; f++) {
-		port = (s->data_block_counter + f) % 8;
+		port = (data_block_counter + f) % 8;
 		b = (u8 *)&buffer[0];
 
 		len = 0;
@@ -330,9 +330,8 @@ void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   __be32 *buffer,
-					   unsigned int data_blocks,
-					   unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
@@ -351,9 +350,8 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   __be32 *buffer,
-					   unsigned int data_blocks,
-					   unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
@@ -367,7 +365,7 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 		pcm_frames = 0;
 	}
 
-	write_midi_messages(s, buffer, data_blocks);
+	write_midi_messages(s, buffer, data_blocks, data_block_counter);
 
 	return pcm_frames;
 }
diff --git a/sound/firewire/fireface/amdtp-ff.c b/sound/firewire/fireface/amdtp-ff.c
index 2938489740b4..71879922ab39 100644
--- a/sound/firewire/fireface/amdtp-ff.c
+++ b/sound/firewire/fireface/amdtp-ff.c
@@ -103,9 +103,8 @@ int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   __be32 *buffer,
-					   unsigned int data_blocks,
-					   unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
@@ -122,9 +121,8 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   __be32 *buffer,
-					   unsigned int data_blocks,
-					   unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 7973dedd31ef..9693f37a0032 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -299,8 +299,8 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct amdtp_motu *p = s->protocol;
 	struct snd_pcm_substream *pcm;
@@ -361,8 +361,8 @@ static void write_sph(struct amdtp_stream *s, __be32 *buffer,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct amdtp_motu *p = (struct amdtp_motu *)s->protocol;
 	struct snd_pcm_substream *pcm;
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index 95fb10b7a737..3a4d1f855f79 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -166,9 +166,8 @@ static void read_status_messages(struct amdtp_stream *s,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   __be32 *buffer,
-					   unsigned int data_blocks,
-					   unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct snd_pcm_substream *pcm;
 
@@ -182,9 +181,8 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   __be32 *buffer,
-					   unsigned int data_blocks,
-					   unsigned int *syt)
+			__be32 *buffer, unsigned int data_blocks,
+			unsigned int data_block_counter, unsigned int *syt)
 {
 	struct snd_pcm_substream *pcm;
 
-- 
cgit v1.2.3-59-g8ed1b


From 860d798c370409efbedf66bae2d41b3cbdf14a79 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:54 +0900
Subject: ALSA: firewire-lib: operate data block counter in top level of
 processing for IT context

In ALSA IEC 61883-1/6 packet streaming engine, two types of data block
counter are supported. This commit applies code refactoring to make it
explicitly for IT context.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 39 ++++++++++++++++++++-------------------
 1 file changed, 20 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 40de9fc9d751..c89083ccd729 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -473,12 +473,12 @@ static inline int queue_in_packet(struct amdtp_stream *s,
 }
 
 static void generate_cip_header(struct amdtp_stream *s, __be32 cip_header[2],
-				unsigned int syt)
+			unsigned int data_block_counter, unsigned int syt)
 {
 	cip_header[0] = cpu_to_be32(READ_ONCE(s->source_node_id_field) |
 				(s->data_block_quadlets << CIP_DBS_SHIFT) |
 				((s->sph << CIP_SPH_SHIFT) & CIP_SPH_MASK) |
-				s->data_block_counter);
+				data_block_counter);
 	cip_header[1] = cpu_to_be32(CIP_EOH |
 			((s->fmt << CIP_FMT_SHIFT) & CIP_FMT_MASK) |
 			((s->ctx_data.rx.fdf << CIP_FDF_SHIFT) & CIP_FDF_MASK) |
@@ -487,8 +487,9 @@ static void generate_cip_header(struct amdtp_stream *s, __be32 cip_header[2],
 
 static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle,
 				struct fw_iso_packet *params,
-				unsigned int data_blocks, unsigned int syt,
-				unsigned int index)
+				unsigned int data_blocks,
+				unsigned int data_block_counter,
+				unsigned int syt, unsigned int index)
 {
 	unsigned int payload_length;
 	__be32 *cip_header;
@@ -496,14 +497,9 @@ static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle,
 	payload_length = data_blocks * sizeof(__be32) * s->data_block_quadlets;
 	params->payload_length = payload_length;
 
-	if (s->flags & CIP_DBC_IS_END_EVENT) {
-		s->data_block_counter =
-				(s->data_block_counter + data_blocks) & 0xff;
-	}
-
 	if (!(s->flags & CIP_NO_HEADER)) {
 		cip_header = (__be32 *)params->header;
-		generate_cip_header(s, cip_header, syt);
+		generate_cip_header(s, cip_header, data_block_counter, syt);
 		params->header_length = 2 * sizeof(__be32);
 		payload_length += params->header_length;
 	} else {
@@ -511,12 +507,7 @@ static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle,
 	}
 
 	trace_amdtp_packet(s, cycle, cip_header, payload_length, data_blocks,
-			   s->data_block_counter, index);
-
-	if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
-		s->data_block_counter =
-				(s->data_block_counter + data_blocks) & 0xff;
-	}
+			   data_block_counter, index);
 }
 
 static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
@@ -709,6 +700,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		u32 cycle;
 		unsigned int syt;
 		unsigned int data_blocks;
+		unsigned int dbc;
 		__be32 *buffer;
 		unsigned int pcm_frames;
 		struct {
@@ -721,11 +713,20 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		syt = calculate_syt(s, cycle);
 		data_blocks = calculate_data_blocks(s, syt);
 		buffer = s->buffer.packets[s->packet_index].buffer;
-		pcm_frames = s->process_data_blocks(s, buffer, data_blocks,
-						s->data_block_counter, &syt);
+		dbc = s->data_block_counter;
+		pcm_frames = s->process_data_blocks(s, buffer, data_blocks, dbc,
+						    &syt);
+
+		if (s->flags & CIP_DBC_IS_END_EVENT)
+			dbc = (dbc + data_blocks) & 0xff;
 
 		build_it_pkt_header(s, cycle, &template.params, data_blocks,
-				    syt, i);
+				    dbc, syt, i);
+
+		if (!(s->flags & CIP_DBC_IS_END_EVENT))
+			dbc = (dbc + data_blocks) & 0xff;
+
+		s->data_block_counter = dbc;
 
 		if (queue_out_packet(s, &template.params) < 0) {
 			cancel_stream(s);
-- 
cgit v1.2.3-59-g8ed1b


From a35463d1f77d9e5244d6f4953d3c7f3a27c7625e Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:55 +0900
Subject: ALSA: firewire-lib: operate data block counter in top level of
 processing for IR context

In ALSA IEC 61883-1/6 packet streaming engine, two types of data block
counter are supported. This commit applies code refactoring to make it
explicitly for IR context.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 53 +++++++++++++++++++++----------------------
 1 file changed, 26 insertions(+), 27 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index c89083ccd729..3435bef97a8b 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -512,13 +512,14 @@ static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle,
 
 static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
 			    unsigned int payload_length,
-			    unsigned int *data_blocks, unsigned int *dbc,
-			    unsigned int *syt)
+			    unsigned int *data_blocks,
+			    unsigned int *data_block_counter, unsigned int *syt)
 {
 	u32 cip_header[2];
 	unsigned int sph;
 	unsigned int fmt;
 	unsigned int fdf;
+	unsigned int dbc;
 	bool lost;
 
 	cip_header[0] = be32_to_cpu(buf[0]);
@@ -570,16 +571,16 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
 	}
 
 	/* Check data block counter continuity */
-	*dbc = cip_header[0] & CIP_DBC_MASK;
+	dbc = cip_header[0] & CIP_DBC_MASK;
 	if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) &&
-	    s->data_block_counter != UINT_MAX)
-		*dbc = s->data_block_counter;
+	    *data_block_counter != UINT_MAX)
+		dbc = *data_block_counter;
 
-	if ((*dbc == 0x00 && (s->flags & CIP_SKIP_DBC_ZERO_CHECK)) ||
-	    s->data_block_counter == UINT_MAX) {
+	if ((dbc == 0x00 && (s->flags & CIP_SKIP_DBC_ZERO_CHECK)) ||
+	    *data_block_counter == UINT_MAX) {
 		lost = false;
 	} else if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
-		lost = *dbc != s->data_block_counter;
+		lost = dbc != *data_block_counter;
 	} else {
 		unsigned int dbc_interval;
 
@@ -588,13 +589,13 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
 		else
 			dbc_interval = *data_blocks;
 
-		lost = *dbc != ((s->data_block_counter + dbc_interval) & 0xff);
+		lost = dbc != ((*data_block_counter + dbc_interval) & 0xff);
 	}
 
 	if (lost) {
 		dev_err(&s->unit->device,
 			"Detect discontinuity of CIP: %02X %02X\n",
-			s->data_block_counter, *dbc);
+			*data_block_counter, dbc);
 		return -EIO;
 	}
 
@@ -606,10 +607,10 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
 static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle,
 			       const __be32 *ctx_header,
 			       unsigned int *payload_length,
-			       unsigned int *data_blocks, unsigned int *syt,
-			       unsigned int index)
+			       unsigned int *data_blocks,
+			       unsigned int *data_block_counter,
+			       unsigned int *syt, unsigned int index)
 {
-	unsigned int dbc;
 	const __be32 *cip_header;
 	int err;
 
@@ -625,7 +626,7 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle,
 	if (!(s->flags & CIP_NO_HEADER)) {
 		cip_header = ctx_header + 2;
 		err = check_cip_header(s, cip_header, *payload_length,
-				       data_blocks, &dbc, syt);
+				       data_blocks, data_block_counter, syt);
 		if (err < 0)
 			return err;
 	} else {
@@ -635,16 +636,12 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle,
 			       s->data_block_quadlets;
 		*syt = 0;
 
-		if (s->data_block_counter != UINT_MAX)
-			dbc = s->data_block_counter;
-		else
-			dbc = 0;
+		if (*data_block_counter == UINT_MAX)
+			*data_block_counter = 0;
 	}
 
-	s->data_block_counter = dbc;
-
 	trace_amdtp_packet(s, cycle, cip_header, *payload_length, *data_blocks,
-			   s->data_block_counter, index);
+			   *data_block_counter, index);
 
 	return err;
 }
@@ -761,6 +758,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		u32 cycle;
 		unsigned int payload_length;
 		unsigned int data_blocks;
+		unsigned int dbc;
 		unsigned int syt;
 		__be32 *buffer;
 		unsigned int pcm_frames = 0;
@@ -769,22 +767,23 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		int err;
 
 		cycle = compute_cycle_count(ctx_header[1]);
+		dbc = s->data_block_counter;
 		err = parse_ir_ctx_header(s, cycle, ctx_header, &payload_length,
-					  &data_blocks, &syt, i);
+					  &data_blocks, &dbc, &syt, i);
 		if (err < 0 && err != -EAGAIN)
 			break;
 
 		if (err >= 0) {
 			buffer = s->buffer.packets[s->packet_index].buffer;
 			pcm_frames = s->process_data_blocks(s, buffer,
-				data_blocks, s->data_block_counter, &syt);
+				data_blocks, dbc, &syt);
 
-			if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
-				s->data_block_counter += data_blocks;
-				s->data_block_counter &= 0xff;
-			}
+			if (!(s->flags & CIP_DBC_IS_END_EVENT))
+				dbc = (dbc + data_blocks) & 0xff;
 		}
 
+		s->data_block_counter = dbc;
+
 		if (queue_in_packet(s, &params) < 0)
 			break;
 
-- 
cgit v1.2.3-59-g8ed1b


From 3baf30532f94030244a31a275b546d042656c535 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:56 +0900
Subject: ALSA: firewire-lib: add syt_override member for some protocols

Some protocols don't use syt field of CIP header to represent
presentation timestamp. For such protocol, ALSA IEC 61883-1/6
packet streaming engine uses a pointer into local variable for
the value of syt to call data block processing layer. However,
it can decide the value when initializing packet streaming
layer.

This commit adds 'syt_override' member for packet streaming
layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c        | 6 ++++++
 sound/firewire/amdtp-stream.h        | 1 +
 sound/firewire/motu/amdtp-motu.c     | 8 +++++++-
 sound/firewire/tascam/amdtp-tascam.c | 8 ++++++--
 4 files changed, 20 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 3435bef97a8b..6242240cd8ee 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -104,6 +104,9 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->fmt = fmt;
 	s->process_data_blocks = process_data_blocks;
 
+	if (dir == AMDTP_OUT_STREAM)
+		s->ctx_data.rx.syt_override = -1;
+
 	return 0;
 }
 EXPORT_SYMBOL(amdtp_stream_init);
@@ -717,6 +720,9 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		if (s->flags & CIP_DBC_IS_END_EVENT)
 			dbc = (dbc + data_blocks) & 0xff;
 
+		if (s->ctx_data.rx.syt_override >= 0)
+			syt = s->ctx_data.rx.syt_override;
+
 		build_it_pkt_header(s, cycle, &template.params, data_blocks,
 				    dbc, syt, i);
 
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 762ac3c7e902..5b9d4212e202 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -133,6 +133,7 @@ struct amdtp_stream {
 
 			// To generate CIP header.
 			unsigned int fdf;
+			int syt_override;
 		} rx;
 	} ctx_data;
 
diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 9693f37a0032..683873699885 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -428,7 +428,13 @@ int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
 		return err;
 
 	s->sph = 1;
-	s->ctx_data.rx.fdf = MOTU_FDF_AM824;
+
+	if (dir == AMDTP_OUT_STREAM) {
+		// Use fixed value for FDF field.
+		s->ctx_data.rx.fdf = MOTU_FDF_AM824;
+		// Not used.
+		s->ctx_data.rx.syt_override = 0xffff;
+	}
 
 	return 0;
 }
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index 3a4d1f855f79..8fba6fb8ba8a 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -220,8 +220,12 @@ int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
 	if (err < 0)
 		return 0;
 
-	/* Use fixed value for FDF field. */
-	s->ctx_data.rx.fdf = 0x00;
+	if (dir == AMDTP_OUT_STREAM) {
+		// Use fixed value for FDF field.
+		s->ctx_data.rx.fdf = 0x00;
+		// Not used.
+		s->ctx_data.rx.syt_override = 0x0000;
+	}
 
 	/* This protocol uses fixed number of data channels for PCM samples. */
 	p = s->protocol;
-- 
cgit v1.2.3-59-g8ed1b


From 600c8018df605a91575aef6811cf927e2d933d30 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:57 +0900
Subject: ALSA: firewire-lib: pass no syt information to data block processing
 layer

In a previous commit, the variable passed from packet streaming layer
for syt variable is useless. This commit obsoletes it.

In my future work, the syt information is passed to data block processing
layer by another way.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-am824.c         |  8 ++++----
 sound/firewire/amdtp-stream.c        |  5 ++---
 sound/firewire/amdtp-stream.h        |  3 +--
 sound/firewire/digi00x/amdtp-dot.c   |  8 ++++----
 sound/firewire/fireface/amdtp-ff.c   |  8 ++++----
 sound/firewire/motu/amdtp-motu.c     | 11 ++++-------
 sound/firewire/tascam/amdtp-tascam.c | 11 ++++-------
 7 files changed, 23 insertions(+), 31 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 03a7e0533131..21068b23d528 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -336,8 +336,8 @@ static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct amdtp_am824 *p = s->protocol;
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
@@ -358,8 +358,8 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct amdtp_am824 *p = s->protocol;
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 6242240cd8ee..c8543cdb3c8c 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -714,8 +714,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		data_blocks = calculate_data_blocks(s, syt);
 		buffer = s->buffer.packets[s->packet_index].buffer;
 		dbc = s->data_block_counter;
-		pcm_frames = s->process_data_blocks(s, buffer, data_blocks, dbc,
-						    &syt);
+		pcm_frames = s->process_data_blocks(s, buffer, data_blocks, dbc);
 
 		if (s->flags & CIP_DBC_IS_END_EVENT)
 			dbc = (dbc + data_blocks) & 0xff;
@@ -782,7 +781,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		if (err >= 0) {
 			buffer = s->buffer.packets[s->packet_index].buffer;
 			pcm_frames = s->process_data_blocks(s, buffer,
-				data_blocks, dbc, &syt);
+							    data_blocks, dbc);
 
 			if (!(s->flags & CIP_DBC_IS_END_EVENT))
 				dbc = (dbc + data_blocks) & 0xff;
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 5b9d4212e202..5996cc60f166 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -99,8 +99,7 @@ typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
 						struct amdtp_stream *s,
 						__be32 *buffer,
 						unsigned int data_blocks,
-						unsigned int data_block_counter,
-						unsigned int *syt);
+						unsigned int data_block_counter);
 struct amdtp_stream {
 	struct fw_unit *unit;
 	enum cip_flags flags;
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
index 5c12973c2652..c296d1017ed3 100644
--- a/sound/firewire/digi00x/amdtp-dot.c
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -330,8 +330,8 @@ void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
@@ -350,8 +350,8 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
diff --git a/sound/firewire/fireface/amdtp-ff.c b/sound/firewire/fireface/amdtp-ff.c
index 71879922ab39..31a60dff94ac 100644
--- a/sound/firewire/fireface/amdtp-ff.c
+++ b/sound/firewire/fireface/amdtp-ff.c
@@ -103,8 +103,8 @@ int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
@@ -121,8 +121,8 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 683873699885..30d5f87119cc 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -299,8 +299,8 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct amdtp_motu *p = s->protocol;
 	struct snd_pcm_substream *pcm;
@@ -361,15 +361,12 @@ static void write_sph(struct amdtp_stream *s, __be32 *buffer,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct amdtp_motu *p = (struct amdtp_motu *)s->protocol;
 	struct snd_pcm_substream *pcm;
 
-	/* Not used. */
-	*syt = 0xffff;
-
 	/* TODO: how to interact control messages between userspace? */
 
 	if (p->midi_ports)
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index 8fba6fb8ba8a..bc1f2d2120b4 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -166,8 +166,8 @@ static void read_status_messages(struct amdtp_stream *s,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct snd_pcm_substream *pcm;
 
@@ -181,14 +181,11 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-			__be32 *buffer, unsigned int data_blocks,
-			unsigned int data_block_counter, unsigned int *syt)
+				__be32 *buffer, unsigned int data_blocks,
+				unsigned int data_block_counter)
 {
 	struct snd_pcm_substream *pcm;
 
-	/* This field is not used. */
-	*syt = 0x0000;
-
 	pcm = READ_ONCE(s->pcm);
 	if (pcm)
 		write_pcm_s32(s, pcm, buffer, data_blocks);
-- 
cgit v1.2.3-59-g8ed1b


From 04130cf8e296bfbd65d2673a5975bd0f200b941d Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:58 +0900
Subject: ALSA: firewire-lib: add list of packet descriptor

In current implementation of ALSA IEC 61883-1/6 packet streaming engine,
16 packets are handled in one interrupt of isochronous context of OHCI
1394.

Overall packet processing runs for each. However, this is not better in
a point to split the processing into several parts.

This commit is an attempt to add intermediate representation for
parameters required for the parts.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 15 ++++++++++++---
 sound/firewire/amdtp-stream.h |  9 +++++++++
 2 files changed, 21 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index c8543cdb3c8c..2bea15151d4a 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -936,6 +936,13 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed)
 	else
 		s->tag = TAG_CIP;
 
+	s->pkt_descs = kcalloc(INTERRUPT_INTERVAL, sizeof(*s->pkt_descs),
+			       GFP_KERNEL);
+	if (!s->pkt_descs) {
+		err = -ENOMEM;
+		goto err_context;
+	}
+
 	s->packet_index = 0;
 	do {
 		struct fw_iso_packet params;
@@ -947,7 +954,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed)
 			err = queue_out_packet(s, &params);
 		}
 		if (err < 0)
-			goto err_context;
+			goto err_pkt_descs;
 	} while (s->packet_index > 0);
 
 	/* NOTE: TAG1 matches CIP. This just affects in stream. */
@@ -958,12 +965,13 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed)
 	s->callbacked = false;
 	err = fw_iso_context_start(s->context, -1, 0, tag);
 	if (err < 0)
-		goto err_context;
+		goto err_pkt_descs;
 
 	mutex_unlock(&s->mutex);
 
 	return 0;
-
+err_pkt_descs:
+	kfree(s->pkt_descs);
 err_context:
 	fw_iso_context_destroy(s->context);
 	s->context = ERR_PTR(-1);
@@ -1059,6 +1067,7 @@ void amdtp_stream_stop(struct amdtp_stream *s)
 	fw_iso_context_destroy(s->context);
 	s->context = ERR_PTR(-1);
 	iso_packets_buffer_destroy(&s->buffer, s->unit);
+	kfree(s->pkt_descs);
 
 	s->callbacked = false;
 
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 5996cc60f166..73c492c9a5d5 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -94,6 +94,14 @@ enum amdtp_stream_direction {
 	AMDTP_IN_STREAM
 };
 
+struct pkt_desc {
+	u32 cycle;
+	u32 syt;
+	unsigned int data_blocks;
+	unsigned int data_block_counter;
+	__be32 *ctx_payload;
+};
+
 struct amdtp_stream;
 typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
 						struct amdtp_stream *s,
@@ -110,6 +118,7 @@ struct amdtp_stream {
 	struct fw_iso_context *context;
 	struct iso_packets_buffer buffer;
 	int packet_index;
+	struct pkt_desc *pkt_descs;
 	int tag;
 	union {
 		struct {
-- 
cgit v1.2.3-59-g8ed1b


From f4f6ae7b7c1fdbaaaecafd183fabcf5314df2b59 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:36:59 +0900
Subject: ALSA: firewire-lib: use packet descriptor for IT context

This commit uses packet descriptor to parse headers of IT context and
generate timing information for ideal sequence.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 68 ++++++++++++++++++++++++++++---------------
 1 file changed, 44 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 2bea15151d4a..463c7a340a45 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -676,6 +676,38 @@ static inline u32 compute_it_cycle(const __be32 ctx_header_tstamp)
 	return increment_cycle_count(cycle, QUEUE_LENGTH);
 }
 
+static void generate_ideal_pkt_descs(struct amdtp_stream *s,
+				     struct pkt_desc *descs,
+				     const __be32 *ctx_header,
+				     unsigned int packets)
+{
+	unsigned int dbc = s->data_block_counter;
+	int i;
+
+	for (i = 0; i < packets; ++i) {
+		struct pkt_desc *desc = descs + i;
+		unsigned int index = (s->packet_index + i) % QUEUE_LENGTH;
+
+		desc->cycle = compute_cycle_count(*ctx_header);
+		desc->syt = calculate_syt(s, desc->cycle);
+		desc->data_blocks = calculate_data_blocks(s, desc->syt);
+
+		if (s->flags & CIP_DBC_IS_END_EVENT)
+			dbc = (dbc + desc->data_blocks) & 0xff;
+
+		desc->data_block_counter = dbc;
+
+		if (!(s->flags & CIP_DBC_IS_END_EVENT))
+			dbc = (dbc + desc->data_blocks) & 0xff;
+
+		desc->ctx_payload = s->buffer.packets[index].buffer;
+
+		++ctx_header;
+	}
+
+	s->data_block_counter = dbc;
+}
+
 static inline void cancel_stream(struct amdtp_stream *s)
 {
 	s->packet_index = -1;
@@ -696,39 +728,29 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 	if (s->packet_index < 0)
 		return;
 
+	generate_ideal_pkt_descs(s, s->pkt_descs, ctx_header, packets);
+
 	for (i = 0; i < packets; ++i) {
-		u32 cycle;
-		unsigned int syt;
-		unsigned int data_blocks;
-		unsigned int dbc;
-		__be32 *buffer;
+		const struct pkt_desc *desc = s->pkt_descs + i;
 		unsigned int pcm_frames;
+		unsigned int syt;
 		struct {
 			struct fw_iso_packet params;
 			__be32 header[IT_PKT_HEADER_SIZE_CIP / sizeof(__be32)];
 		} template = { {0}, {0} };
 		struct snd_pcm_substream *pcm;
 
-		cycle = compute_it_cycle(*ctx_header);
-		syt = calculate_syt(s, cycle);
-		data_blocks = calculate_data_blocks(s, syt);
-		buffer = s->buffer.packets[s->packet_index].buffer;
-		dbc = s->data_block_counter;
-		pcm_frames = s->process_data_blocks(s, buffer, data_blocks, dbc);
+		pcm_frames = s->process_data_blocks(s, desc->ctx_payload,
+				desc->data_blocks, desc->data_block_counter);
 
-		if (s->flags & CIP_DBC_IS_END_EVENT)
-			dbc = (dbc + data_blocks) & 0xff;
-
-		if (s->ctx_data.rx.syt_override >= 0)
+		if (s->ctx_data.rx.syt_override < 0)
+			syt = desc->syt;
+		else
 			syt = s->ctx_data.rx.syt_override;
 
-		build_it_pkt_header(s, cycle, &template.params, data_blocks,
-				    dbc, syt, i);
-
-		if (!(s->flags & CIP_DBC_IS_END_EVENT))
-			dbc = (dbc + data_blocks) & 0xff;
-
-		s->data_block_counter = dbc;
+		build_it_pkt_header(s, desc->cycle, &template.params,
+				    desc->data_blocks, desc->data_block_counter,
+				    syt, i);
 
 		if (queue_out_packet(s, &template.params) < 0) {
 			cancel_stream(s);
@@ -738,8 +760,6 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		pcm = READ_ONCE(s->pcm);
 		if (pcm && pcm_frames > 0)
 			update_pcm_pointers(s, pcm, pcm_frames);
-
-		++ctx_header;
 	}
 
 	fw_iso_context_queue_flush(s->context);
-- 
cgit v1.2.3-59-g8ed1b


From 753e717986c24b60962367708479555212730b9f Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:00 +0900
Subject: ALSA: firewire-lib: use packet descriptor for IR context

This commit uses packet descriptor to parse headers of IR context and
parse timing information for sequence transferred by target device.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 96 ++++++++++++++++++++++++++++---------------
 1 file changed, 63 insertions(+), 33 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 463c7a340a45..68502a8864b9 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -602,6 +602,8 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf,
 		return -EIO;
 	}
 
+	*data_block_counter = dbc;
+
 	*syt = cip_header[1] & CIP_SYT_MASK;
 
 	return 0;
@@ -676,6 +678,48 @@ static inline u32 compute_it_cycle(const __be32 ctx_header_tstamp)
 	return increment_cycle_count(cycle, QUEUE_LENGTH);
 }
 
+static int generate_device_pkt_descs(struct amdtp_stream *s,
+				     struct pkt_desc *descs,
+				     const __be32 *ctx_header,
+				     unsigned int packets)
+{
+	unsigned int dbc = s->data_block_counter;
+	int i;
+	int err;
+
+	for (i = 0; i < packets; ++i) {
+		struct pkt_desc *desc = descs + i;
+		unsigned int index = (s->packet_index + i) % QUEUE_LENGTH;
+		unsigned int cycle;
+		unsigned int payload_length;
+		unsigned int data_blocks;
+		unsigned int syt;
+
+		cycle = compute_cycle_count(ctx_header[1]);
+
+		err = parse_ir_ctx_header(s, cycle, ctx_header, &payload_length,
+					  &data_blocks, &dbc, &syt, i);
+		if (err < 0)
+			return err;
+
+		desc->cycle = cycle;
+		desc->syt = syt;
+		desc->data_blocks = data_blocks;
+		desc->data_block_counter = dbc;
+		desc->ctx_payload = s->buffer.packets[index].buffer;
+
+		if (!(s->flags & CIP_DBC_IS_END_EVENT))
+			dbc = (dbc + desc->data_blocks) & 0xff;
+
+		ctx_header +=
+			s->ctx_data.tx.ctx_header_size / sizeof(*ctx_header);
+	}
+
+	s->data_block_counter = dbc;
+
+	return 0;
+}
+
 static void generate_ideal_pkt_descs(struct amdtp_stream *s,
 				     struct pkt_desc *descs,
 				     const __be32 *ctx_header,
@@ -770,8 +814,10 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 			       void *private_data)
 {
 	struct amdtp_stream *s = private_data;
-	unsigned int i, packets;
+	unsigned int packets;
 	__be32 *ctx_header = header;
+	int i;
+	int err;
 
 	if (s->packet_index < 0)
 		return;
@@ -779,50 +825,34 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 	// The number of packets in buffer.
 	packets = header_length / s->ctx_data.tx.ctx_header_size;
 
+	err = generate_device_pkt_descs(s, s->pkt_descs, ctx_header, packets);
+	if (err < 0) {
+		if (err != -EAGAIN) {
+			cancel_stream(s);
+			return;
+		}
+	}
+
 	for (i = 0; i < packets; i++) {
-		u32 cycle;
-		unsigned int payload_length;
-		unsigned int data_blocks;
-		unsigned int dbc;
-		unsigned int syt;
-		__be32 *buffer;
+		const struct pkt_desc *desc = s->pkt_descs;
 		unsigned int pcm_frames = 0;
 		struct fw_iso_packet params = {0};
 		struct snd_pcm_substream *pcm;
-		int err;
-
-		cycle = compute_cycle_count(ctx_header[1]);
-		dbc = s->data_block_counter;
-		err = parse_ir_ctx_header(s, cycle, ctx_header, &payload_length,
-					  &data_blocks, &dbc, &syt, i);
-		if (err < 0 && err != -EAGAIN)
-			break;
 
 		if (err >= 0) {
-			buffer = s->buffer.packets[s->packet_index].buffer;
-			pcm_frames = s->process_data_blocks(s, buffer,
-							    data_blocks, dbc);
-
-			if (!(s->flags & CIP_DBC_IS_END_EVENT))
-				dbc = (dbc + data_blocks) & 0xff;
+			pcm_frames = s->process_data_blocks(s,
+					desc->ctx_payload, desc->data_blocks,
+					desc->data_block_counter);
 		}
 
-		s->data_block_counter = dbc;
-
-		if (queue_in_packet(s, &params) < 0)
-			break;
+		if (queue_in_packet(s, &params) < 0) {
+			cancel_stream(s);
+			return;
+		}
 
 		pcm = READ_ONCE(s->pcm);
 		if (pcm && pcm_frames > 0)
 			update_pcm_pointers(s, pcm, pcm_frames);
-
-		ctx_header += s->ctx_data.tx.ctx_header_size / sizeof(*ctx_header);
-	}
-
-	/* Queueing error or detecting invalid payload. */
-	if (i < packets) {
-		cancel_stream(s);
-		return;
 	}
 
 	fw_iso_context_queue_flush(s->context);
-- 
cgit v1.2.3-59-g8ed1b


From 5e2ece0fdceb02f6c47f53bef309395701a65393 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:01 +0900
Subject: ALSA: firewire-lib: code refactoring to process PCM substream

This is code refactoring to separate PCM substream processing from packet
queueing.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 45 ++++++++++++++++++++++++-------------------
 1 file changed, 25 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 68502a8864b9..99b89bd70bc5 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -776,16 +776,24 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 
 	for (i = 0; i < packets; ++i) {
 		const struct pkt_desc *desc = s->pkt_descs + i;
+		struct snd_pcm_substream *pcm;
 		unsigned int pcm_frames;
+
+		pcm_frames = s->process_data_blocks(s, desc->ctx_payload,
+				desc->data_blocks, desc->data_block_counter);
+
+		pcm = READ_ONCE(s->pcm);
+		if (pcm && pcm_frames > 0)
+			update_pcm_pointers(s, pcm, pcm_frames);
+	}
+
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = s->pkt_descs + i;
 		unsigned int syt;
 		struct {
 			struct fw_iso_packet params;
 			__be32 header[IT_PKT_HEADER_SIZE_CIP / sizeof(__be32)];
 		} template = { {0}, {0} };
-		struct snd_pcm_substream *pcm;
-
-		pcm_frames = s->process_data_blocks(s, desc->ctx_payload,
-				desc->data_blocks, desc->data_block_counter);
 
 		if (s->ctx_data.rx.syt_override < 0)
 			syt = desc->syt;
@@ -800,10 +808,6 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 			cancel_stream(s);
 			return;
 		}
-
-		pcm = READ_ONCE(s->pcm);
-		if (pcm && pcm_frames > 0)
-			update_pcm_pointers(s, pcm, pcm_frames);
 	}
 
 	fw_iso_context_queue_flush(s->context);
@@ -831,28 +835,29 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 			cancel_stream(s);
 			return;
 		}
-	}
-
-	for (i = 0; i < packets; i++) {
-		const struct pkt_desc *desc = s->pkt_descs;
-		unsigned int pcm_frames = 0;
-		struct fw_iso_packet params = {0};
-		struct snd_pcm_substream *pcm;
+	} else {
+		for (i = 0; i < packets; ++i) {
+			const struct pkt_desc *desc = s->pkt_descs;
+			struct snd_pcm_substream *pcm;
+			unsigned int pcm_frames;
 
-		if (err >= 0) {
 			pcm_frames = s->process_data_blocks(s,
 					desc->ctx_payload, desc->data_blocks,
 					desc->data_block_counter);
+
+			pcm = READ_ONCE(s->pcm);
+			if (pcm && pcm_frames > 0)
+				update_pcm_pointers(s, pcm, pcm_frames);
 		}
+	}
+
+	for (i = 0; i < packets; ++i) {
+		struct fw_iso_packet params = {0};
 
 		if (queue_in_packet(s, &params) < 0) {
 			cancel_stream(s);
 			return;
 		}
-
-		pcm = READ_ONCE(s->pcm);
-		if (pcm && pcm_frames > 0)
-			update_pcm_pointers(s, pcm, pcm_frames);
 	}
 
 	fw_iso_context_queue_flush(s->context);
-- 
cgit v1.2.3-59-g8ed1b


From 0f5cfcb24d9c931e4571e9995bbfc08cc76d5d4a Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:02 +0900
Subject: ALSA: firewire-lib: code refactoring to process context payloads

This is code refactoring for common processing for payloads of
ishocornous context.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 47 ++++++++++++++++++++-----------------------
 1 file changed, 22 insertions(+), 25 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 99b89bd70bc5..573265113a6f 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -760,22 +760,14 @@ static inline void cancel_stream(struct amdtp_stream *s)
 	WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN);
 }
 
-static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
-				size_t header_length, void *header,
-				void *private_data)
+static void process_ctx_payloads(struct amdtp_stream *s,
+				 const struct pkt_desc *descs,
+				 unsigned int packets)
 {
-	struct amdtp_stream *s = private_data;
-	const __be32 *ctx_header = header;
-	unsigned int packets = header_length / sizeof(*ctx_header);
 	int i;
 
-	if (s->packet_index < 0)
-		return;
-
-	generate_ideal_pkt_descs(s, s->pkt_descs, ctx_header, packets);
-
 	for (i = 0; i < packets; ++i) {
-		const struct pkt_desc *desc = s->pkt_descs + i;
+		const struct pkt_desc *desc = descs + i;
 		struct snd_pcm_substream *pcm;
 		unsigned int pcm_frames;
 
@@ -786,6 +778,23 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 		if (pcm && pcm_frames > 0)
 			update_pcm_pointers(s, pcm, pcm_frames);
 	}
+}
+
+static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
+				size_t header_length, void *header,
+				void *private_data)
+{
+	struct amdtp_stream *s = private_data;
+	const __be32 *ctx_header = header;
+	unsigned int packets = header_length / sizeof(*ctx_header);
+	int i;
+
+	if (s->packet_index < 0)
+		return;
+
+	generate_ideal_pkt_descs(s, s->pkt_descs, ctx_header, packets);
+
+	process_ctx_payloads(s, s->pkt_descs, packets);
 
 	for (i = 0; i < packets; ++i) {
 		const struct pkt_desc *desc = s->pkt_descs + i;
@@ -836,19 +845,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 			return;
 		}
 	} else {
-		for (i = 0; i < packets; ++i) {
-			const struct pkt_desc *desc = s->pkt_descs;
-			struct snd_pcm_substream *pcm;
-			unsigned int pcm_frames;
-
-			pcm_frames = s->process_data_blocks(s,
-					desc->ctx_payload, desc->data_blocks,
-					desc->data_block_counter);
-
-			pcm = READ_ONCE(s->pcm);
-			if (pcm && pcm_frames > 0)
-				update_pcm_pointers(s, pcm, pcm_frames);
-		}
+		process_ctx_payloads(s, s->pkt_descs, packets);
 	}
 
 	for (i = 0; i < packets; ++i) {
-- 
cgit v1.2.3-59-g8ed1b


From d2c104a3426be9991b35c65f0f260a107c4b2942 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:03 +0900
Subject: ALSA: firewire-lib: pass packet descriptor to data block processing
 layer

This commit changes signature of callback function to call data block
processing layer with packet descriptor. At present, the layer is called
per packet.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-am824.c         | 32 ++++++++++++----------
 sound/firewire/amdtp-stream.c        |  6 ++--
 sound/firewire/amdtp-stream.h        |  5 ++--
 sound/firewire/digi00x/amdtp-dot.c   | 27 ++++++++----------
 sound/firewire/fireface/amdtp-ff.c   | 23 ++++++++--------
 sound/firewire/motu/amdtp-motu.c     | 53 ++++++++++++++++++++----------------
 sound/firewire/tascam/amdtp-tascam.c | 39 ++++++++++++++------------
 7 files changed, 97 insertions(+), 88 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 21068b23d528..ff089ffa3374 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -336,44 +336,46 @@ static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
 	struct amdtp_am824 *p = s->protocol;
-	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
 
 	if (pcm) {
-		write_pcm_s32(s, pcm, buffer, data_blocks);
-		pcm_frames = data_blocks * p->frame_multiplier;
+		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = desc->data_blocks * p->frame_multiplier;
 	} else {
-		write_pcm_silence(s, buffer, data_blocks);
+		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
 		pcm_frames = 0;
 	}
 
-	if (p->midi_ports)
-		write_midi_messages(s, buffer, data_blocks, data_block_counter);
+	if (p->midi_ports) {
+		write_midi_messages(s, desc->ctx_payload, desc->data_blocks,
+				    desc->data_block_counter);
+	}
 
 	return pcm_frames;
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
 	struct amdtp_am824 *p = s->protocol;
-	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
 
 	if (pcm) {
-		read_pcm_s32(s, pcm, buffer, data_blocks);
-		pcm_frames = data_blocks * p->frame_multiplier;
+		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = desc->data_blocks * p->frame_multiplier;
 	} else {
 		pcm_frames = 0;
 	}
 
-	if (p->midi_ports)
-		read_midi_messages(s, buffer, data_blocks, data_block_counter);
+	if (p->midi_ports) {
+		read_midi_messages(s, desc->ctx_payload, desc->data_blocks,
+				   desc->data_block_counter);
+	}
 
 	return pcm_frames;
 }
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 573265113a6f..db2feb68105c 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -768,13 +768,11 @@ static void process_ctx_payloads(struct amdtp_stream *s,
 
 	for (i = 0; i < packets; ++i) {
 		const struct pkt_desc *desc = descs + i;
-		struct snd_pcm_substream *pcm;
+		struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 		unsigned int pcm_frames;
 
-		pcm_frames = s->process_data_blocks(s, desc->ctx_payload,
-				desc->data_blocks, desc->data_block_counter);
+		pcm_frames = s->process_data_blocks(s, desc, pcm);
 
-		pcm = READ_ONCE(s->pcm);
 		if (pcm && pcm_frames > 0)
 			update_pcm_pointers(s, pcm, pcm_frames);
 	}
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 73c492c9a5d5..0e5b85100a04 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -105,9 +105,8 @@ struct pkt_desc {
 struct amdtp_stream;
 typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
 						struct amdtp_stream *s,
-						__be32 *buffer,
-						unsigned int data_blocks,
-						unsigned int data_block_counter);
+						const struct pkt_desc *desc,
+						struct snd_pcm_substream *pcm);
 struct amdtp_stream {
 	struct fw_unit *unit;
 	enum cip_flags flags;
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
index c296d1017ed3..83ac4b37f26d 100644
--- a/sound/firewire/digi00x/amdtp-dot.c
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -330,42 +330,39 @@ void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
-	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
 
-	pcm = READ_ONCE(s->pcm);
 	if (pcm) {
-		read_pcm_s32(s, pcm, buffer, data_blocks);
-		pcm_frames = data_blocks;
+		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = desc->data_blocks;
 	} else {
 		pcm_frames = 0;
 	}
 
-	read_midi_messages(s, buffer, data_blocks);
+	read_midi_messages(s, desc->ctx_payload, desc->data_blocks);
 
 	return pcm_frames;
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
-	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
 
-	pcm = READ_ONCE(s->pcm);
 	if (pcm) {
-		write_pcm_s32(s, pcm, buffer, data_blocks);
-		pcm_frames = data_blocks;
+		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = desc->data_blocks;
 	} else {
-		write_pcm_silence(s, buffer, data_blocks);
+		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
 		pcm_frames = 0;
 	}
 
-	write_midi_messages(s, buffer, data_blocks, data_block_counter);
+	write_midi_messages(s, desc->ctx_payload, desc->data_blocks,
+			    desc->data_block_counter);
 
 	return pcm_frames;
 }
diff --git a/sound/firewire/fireface/amdtp-ff.c b/sound/firewire/fireface/amdtp-ff.c
index 31a60dff94ac..c36232fc4d3e 100644
--- a/sound/firewire/fireface/amdtp-ff.c
+++ b/sound/firewire/fireface/amdtp-ff.c
@@ -103,17 +103,18 @@ int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
-	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
 
 	if (pcm) {
-		write_pcm_s32(s, pcm, (__le32 *)buffer, data_blocks);
-		pcm_frames = data_blocks;
+		write_pcm_s32(s, pcm, (__le32 *)desc->ctx_payload,
+			      desc->data_blocks);
+		pcm_frames = desc->data_blocks;
 	} else {
-		write_pcm_silence(s, (__le32 *)buffer, data_blocks);
+		write_pcm_silence(s, (__le32 *)desc->ctx_payload,
+				  desc->data_blocks);
 		pcm_frames = 0;
 	}
 
@@ -121,15 +122,15 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
-	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 	unsigned int pcm_frames;
 
 	if (pcm) {
-		read_pcm_s32(s, pcm, (__le32 *)buffer, data_blocks);
-		pcm_frames = data_blocks;
+		read_pcm_s32(s, pcm, (__le32 *)desc->ctx_payload,
+			     desc->data_blocks);
+		pcm_frames = desc->data_blocks;
 	} else {
 		pcm_frames = 0;
 	}
diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 30d5f87119cc..36ee2c1dd667 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -299,23 +299,27 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
 	struct amdtp_motu *p = s->protocol;
-	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
 
-	trace_data_block_sph(s, data_blocks, buffer);
-	trace_data_block_message(s, data_blocks, buffer);
+	trace_data_block_sph(s, desc->data_blocks, desc->ctx_payload);
+	trace_data_block_message(s, desc->data_blocks, desc->ctx_payload);
 
 	if (p->midi_ports)
-		read_midi_messages(s, buffer, data_blocks);
+		read_midi_messages(s, desc->ctx_payload, desc->data_blocks);
 
-	pcm = READ_ONCE(s->pcm);
-	if (data_blocks > 0 && pcm)
-		read_pcm_s32(s, pcm->runtime, buffer, data_blocks);
+	if (pcm) {
+		read_pcm_s32(s, pcm->runtime, desc->ctx_payload,
+			     desc->data_blocks);
+		pcm_frames = desc->data_blocks;
+	} else {
+		pcm_frames = 0;
+	}
 
-	return data_blocks;
+	return pcm_frames;
 }
 
 static inline void compute_next_elapse_from_start(struct amdtp_motu *p)
@@ -361,29 +365,32 @@ static void write_sph(struct amdtp_stream *s, __be32 *buffer,
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
 	struct amdtp_motu *p = (struct amdtp_motu *)s->protocol;
-	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
 
 	/* TODO: how to interact control messages between userspace? */
 
 	if (p->midi_ports)
-		write_midi_messages(s, buffer, data_blocks);
+		write_midi_messages(s, desc->ctx_payload, desc->data_blocks);
 
-	pcm = READ_ONCE(s->pcm);
-	if (pcm)
-		write_pcm_s32(s, pcm->runtime, buffer, data_blocks);
-	else
-		write_pcm_silence(s, buffer, data_blocks);
+	if (pcm) {
+		write_pcm_s32(s, pcm->runtime, desc->ctx_payload,
+			      desc->data_blocks);
+		pcm_frames = desc->data_blocks;
+	} else {
+		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = 0;
+	}
 
-	write_sph(s, buffer, data_blocks);
+	write_sph(s, desc->ctx_payload, desc->data_blocks);
 
-	trace_data_block_sph(s, data_blocks, buffer);
-	trace_data_block_message(s, data_blocks, buffer);
+	trace_data_block_sph(s, desc->data_blocks, desc->ctx_payload);
+	trace_data_block_message(s, desc->data_blocks, desc->ctx_payload);
 
-	return data_blocks;
+	return pcm_frames;
 }
 
 int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index bc1f2d2120b4..970b1c4a8ea6 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -166,33 +166,38 @@ static void read_status_messages(struct amdtp_stream *s,
 }
 
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
-	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
 
-	pcm = READ_ONCE(s->pcm);
-	if (data_blocks > 0 && pcm)
-		read_pcm_s32(s, pcm, buffer, data_blocks);
+	if (pcm) {
+		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = desc->data_blocks;
+	} else {
+		pcm_frames = 0;
+	}
 
-	read_status_messages(s, buffer, data_blocks);
+	read_status_messages(s, desc->ctx_payload, desc->data_blocks);
 
-	return data_blocks;
+	return pcm_frames;
 }
 
 static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int data_blocks,
-				unsigned int data_block_counter)
+					   const struct pkt_desc *desc,
+					   struct snd_pcm_substream *pcm)
 {
-	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
 
-	pcm = READ_ONCE(s->pcm);
-	if (pcm)
-		write_pcm_s32(s, pcm, buffer, data_blocks);
-	else
-		write_pcm_silence(s, buffer, data_blocks);
+	if (pcm) {
+		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = desc->data_blocks;
+	} else {
+		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
+		pcm_frames = 0;
+	}
 
-	return data_blocks;
+	return pcm_frames;
 }
 
 int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
-- 
cgit v1.2.3-59-g8ed1b


From 9fc90644cae9c0abfcce0576ed0a14cca67f69b9 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:04 +0900
Subject: ALSA: firewire-lib: code refactoring for AM824 data block processing
 layer

This is code refactoring for AM824 data block processing layer so that
it can receive list of packet descriptor.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-am824.c | 51 ++++++++++++++++++++++++++------------------
 1 file changed, 30 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index ff089ffa3374..5d2952a1a9b2 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -146,19 +146,24 @@ void amdtp_am824_set_midi_position(struct amdtp_stream *s,
 }
 EXPORT_SYMBOL_GPL(amdtp_am824_set_midi_position);
 
-static void write_pcm_s32(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames,
+			  unsigned int pcm_frames)
 {
 	struct amdtp_am824 *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	const u32 *src;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	for (i = 0; i < frames; ++i) {
 		for (c = 0; c < channels; ++c) {
@@ -172,19 +177,24 @@ static void write_pcm_s32(struct amdtp_stream *s,
 	}
 }
 
-static void read_pcm_s32(struct amdtp_stream *s,
-			 struct snd_pcm_substream *pcm,
-			 __be32 *buffer, unsigned int frames)
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			 __be32 *buffer, unsigned int frames,
+			 unsigned int pcm_frames)
 {
 	struct amdtp_am824 *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	u32 *dst;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	dst  = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	for (i = 0; i < frames; ++i) {
 		for (c = 0; c < channels; ++c) {
@@ -340,14 +350,14 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 					   struct snd_pcm_substream *pcm)
 {
 	struct amdtp_am824 *p = s->protocol;
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
-		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			      pcm_frames);
 		pcm_frames = desc->data_blocks * p->frame_multiplier;
 	} else {
 		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
-		pcm_frames = 0;
 	}
 
 	if (p->midi_ports) {
@@ -363,13 +373,12 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 					   struct snd_pcm_substream *pcm)
 {
 	struct amdtp_am824 *p = s->protocol;
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
-		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			     pcm_frames);
 		pcm_frames = desc->data_blocks * p->frame_multiplier;
-	} else {
-		pcm_frames = 0;
 	}
 
 	if (p->midi_ports) {
-- 
cgit v1.2.3-59-g8ed1b


From bb473966774a79ae9a826f721059496d9b393acc Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:05 +0900
Subject: ALSA: firewire-digi00x: code refactoring for DOT data block
 processing layer

This is code refactoring for DOT data block processing layer so that
it can receive list of packet descriptor.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/digi00x/amdtp-dot.c | 45 ++++++++++++++++++++++++--------------
 1 file changed, 28 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
index 83ac4b37f26d..cef5caf97236 100644
--- a/sound/firewire/digi00x/amdtp-dot.c
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -143,17 +143,23 @@ int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
 }
 
 static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
+			  __be32 *buffer, unsigned int frames,
+			  unsigned int pcm_frames)
 {
 	struct amdtp_dot *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	const u32 *src;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	buffer++;
 	for (i = 0; i < frames; ++i) {
@@ -169,17 +175,23 @@ static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
 }
 
 static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
-			 __be32 *buffer, unsigned int frames)
+			 __be32 *buffer, unsigned int frames,
+			 unsigned int pcm_frames)
 {
 	struct amdtp_dot *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	u32 *dst;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	dst  = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	buffer++;
 	for (i = 0; i < frames; ++i) {
@@ -333,13 +345,12 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 					   const struct pkt_desc *desc,
 					   struct snd_pcm_substream *pcm)
 {
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
-		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			     pcm_frames);
 		pcm_frames = desc->data_blocks;
-	} else {
-		pcm_frames = 0;
 	}
 
 	read_midi_messages(s, desc->ctx_payload, desc->data_blocks);
@@ -351,14 +362,14 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 					   const struct pkt_desc *desc,
 					   struct snd_pcm_substream *pcm)
 {
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
-		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
+		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			      pcm_frames);
 		pcm_frames = desc->data_blocks;
 	} else {
 		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
-		pcm_frames = 0;
 	}
 
 	write_midi_messages(s, desc->ctx_payload, desc->data_blocks,
-- 
cgit v1.2.3-59-g8ed1b


From 806653b70ee21b374d2844d144609b2e0070a86b Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:06 +0900
Subject: ALSA: firewire-tascam: code refactoring for TASCAM data block
 processing layer

This is code refactoring for TASCAM data block processing layer so that
it can receive list of packet descriptor.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/tascam/amdtp-tascam.c | 55 +++++++++++++++++++++---------------
 1 file changed, 32 insertions(+), 23 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index 970b1c4a8ea6..3c9abf422f24 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -32,19 +32,24 @@ int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate)
 	return amdtp_stream_set_parameters(s, rate, data_channels);
 }
 
-static void write_pcm_s32(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames,
+			  unsigned int pcm_frames)
 {
 	struct amdtp_tscm *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	const u32 *src;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	for (i = 0; i < frames; ++i) {
 		for (c = 0; c < channels; ++c) {
@@ -57,19 +62,24 @@ static void write_pcm_s32(struct amdtp_stream *s,
 	}
 }
 
-static void read_pcm_s32(struct amdtp_stream *s,
-			 struct snd_pcm_substream *pcm,
-			 __be32 *buffer, unsigned int frames)
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			 __be32 *buffer, unsigned int frames,
+			 unsigned int pcm_frames)
 {
 	struct amdtp_tscm *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	u32 *dst;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	dst  = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	/* The first data channel is for event counter. */
 	buffer += 1;
@@ -169,13 +179,12 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 					   const struct pkt_desc *desc,
 					   struct snd_pcm_substream *pcm)
 {
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
-		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
-		pcm_frames = desc->data_blocks;
-	} else {
-		pcm_frames = 0;
+		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			     pcm_frames);
+		pcm_frames += desc->data_blocks;
 	}
 
 	read_status_messages(s, desc->ctx_payload, desc->data_blocks);
@@ -187,14 +196,14 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 					   const struct pkt_desc *desc,
 					   struct snd_pcm_substream *pcm)
 {
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
-		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks);
-		pcm_frames = desc->data_blocks;
+		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			      pcm_frames);
+		pcm_frames += desc->data_blocks;
 	} else {
 		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
-		pcm_frames = 0;
 	}
 
 	return pcm_frames;
-- 
cgit v1.2.3-59-g8ed1b


From 00d004db6493c62b8bbd7073a9b3b7290cbe0cea Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:07 +0900
Subject: ALSA: firewire-motu: code refactoring for MOTU data block processing
 layer

This is code refactoring for MOTU data block processing layer so that
it can receive list of packet descriptor.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/motu/amdtp-motu.c | 57 +++++++++++++++++++++++-----------------
 1 file changed, 33 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 36ee2c1dd667..21c6bfdf47c0 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -117,19 +117,25 @@ int amdtp_motu_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	return 0;
 }
 
-static void read_pcm_s32(struct amdtp_stream *s,
-			 struct snd_pcm_runtime *runtime,
-			 __be32 *buffer, unsigned int data_blocks)
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			 __be32 *buffer, unsigned int data_blocks,
+			 unsigned int pcm_frames)
 {
 	struct amdtp_motu *p = s->protocol;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int channels = p->pcm_chunks;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	u8 *byte;
 	u32 *dst;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_chunks;
 	dst = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	for (i = 0; i < data_blocks; ++i) {
 		byte = (u8 *)buffer + p->pcm_byte_offset;
@@ -147,19 +153,25 @@ static void read_pcm_s32(struct amdtp_stream *s,
 	}
 }
 
-static void write_pcm_s32(struct amdtp_stream *s,
-			  struct snd_pcm_runtime *runtime,
-			  __be32 *buffer, unsigned int data_blocks)
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int data_blocks,
+			  unsigned int pcm_frames)
 {
 	struct amdtp_motu *p = s->protocol;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int channels = p->pcm_chunks;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	u8 *byte;
 	const u32 *src;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_chunks;
 	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	for (i = 0; i < data_blocks; ++i) {
 		byte = (u8 *)buffer + p->pcm_byte_offset;
@@ -303,7 +315,7 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 					   struct snd_pcm_substream *pcm)
 {
 	struct amdtp_motu *p = s->protocol;
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	trace_data_block_sph(s, desc->data_blocks, desc->ctx_payload);
 	trace_data_block_message(s, desc->data_blocks, desc->ctx_payload);
@@ -312,11 +324,9 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 		read_midi_messages(s, desc->ctx_payload, desc->data_blocks);
 
 	if (pcm) {
-		read_pcm_s32(s, pcm->runtime, desc->ctx_payload,
-			     desc->data_blocks);
+		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			     pcm_frames);
 		pcm_frames = desc->data_blocks;
-	} else {
-		pcm_frames = 0;
 	}
 
 	return pcm_frames;
@@ -368,8 +378,8 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 					   const struct pkt_desc *desc,
 					   struct snd_pcm_substream *pcm)
 {
-	struct amdtp_motu *p = (struct amdtp_motu *)s->protocol;
-	unsigned int pcm_frames;
+	struct amdtp_motu *p = s->protocol;
+	unsigned int pcm_frames = 0;
 
 	/* TODO: how to interact control messages between userspace? */
 
@@ -377,12 +387,11 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 		write_midi_messages(s, desc->ctx_payload, desc->data_blocks);
 
 	if (pcm) {
-		write_pcm_s32(s, pcm->runtime, desc->ctx_payload,
-			      desc->data_blocks);
+		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
+			      pcm_frames);
 		pcm_frames = desc->data_blocks;
 	} else {
 		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
-		pcm_frames = 0;
 	}
 
 	write_sph(s, desc->ctx_payload, desc->data_blocks);
-- 
cgit v1.2.3-59-g8ed1b


From 4731c672c930e490515f1747fb34d56d61ae57ac Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:08 +0900
Subject: ALSA: fireface: code refactoring for FF data block processing layer

This is code refactoring for FF data block processing layer so that
it can receive list of packet descriptor.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/fireface/amdtp-ff.c | 49 ++++++++++++++++++++++----------------
 1 file changed, 28 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/fireface/amdtp-ff.c b/sound/firewire/fireface/amdtp-ff.c
index c36232fc4d3e..cb2073cf146b 100644
--- a/sound/firewire/fireface/amdtp-ff.c
+++ b/sound/firewire/fireface/amdtp-ff.c
@@ -27,19 +27,24 @@ int amdtp_ff_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	return amdtp_stream_set_parameters(s, rate, data_channels);
 }
 
-static void write_pcm_s32(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __le32 *buffer, unsigned int frames)
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __le32 *buffer, unsigned int frames,
+			  unsigned int pcm_frames)
 {
 	struct amdtp_ff *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	const u32 *src;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	for (i = 0; i < frames; ++i) {
 		for (c = 0; c < channels; ++c) {
@@ -52,19 +57,24 @@ static void write_pcm_s32(struct amdtp_stream *s,
 	}
 }
 
-static void read_pcm_s32(struct amdtp_stream *s,
-			 struct snd_pcm_substream *pcm,
-			 __le32 *buffer, unsigned int frames)
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			 __le32 *buffer, unsigned int frames,
+			 unsigned int pcm_frames)
 {
 	struct amdtp_ff *p = s->protocol;
+	unsigned int channels = p->pcm_channels;
 	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
+	unsigned int pcm_buffer_pointer;
+	int remaining_frames;
 	u32 *dst;
+	int i, c;
+
+	pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames;
+	pcm_buffer_pointer %= runtime->buffer_size;
 
-	channels = p->pcm_channels;
 	dst  = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+				frames_to_bytes(runtime, pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - pcm_buffer_pointer;
 
 	for (i = 0; i < frames; ++i) {
 		for (c = 0; c < channels; ++c) {
@@ -106,16 +116,15 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 					   const struct pkt_desc *desc,
 					   struct snd_pcm_substream *pcm)
 {
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
 		write_pcm_s32(s, pcm, (__le32 *)desc->ctx_payload,
-			      desc->data_blocks);
+			      desc->data_blocks, pcm_frames);
 		pcm_frames = desc->data_blocks;
 	} else {
 		write_pcm_silence(s, (__le32 *)desc->ctx_payload,
 				  desc->data_blocks);
-		pcm_frames = 0;
 	}
 
 	return pcm_frames;
@@ -125,14 +134,12 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 					   const struct pkt_desc *desc,
 					   struct snd_pcm_substream *pcm)
 {
-	unsigned int pcm_frames;
+	unsigned int pcm_frames = 0;
 
 	if (pcm) {
 		read_pcm_s32(s, pcm, (__le32 *)desc->ctx_payload,
-			     desc->data_blocks);
+			     desc->data_blocks, pcm_frames);
 		pcm_frames = desc->data_blocks;
-	} else {
-		pcm_frames = 0;
 	}
 
 	return pcm_frames;
-- 
cgit v1.2.3-59-g8ed1b


From 9a738ad1b1a64ef6d8271d8d42d715b75db6eb0d Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:09 +0900
Subject: ALSA: firewire-lib: process payload of isoc context according to
 packet descriptors

This commit changes each of data block processing layer so that it
receives list of packet descriptor.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-am824.c         | 71 +++++++++++++++++++--------------
 sound/firewire/amdtp-stream.c        | 25 +++++-------
 sound/firewire/amdtp-stream.h        |  7 ++--
 sound/firewire/digi00x/amdtp-dot.c   | 66 +++++++++++++++++++------------
 sound/firewire/fireface/amdtp-ff.c   | 55 ++++++++++++++++----------
 sound/firewire/motu/amdtp-motu.c     | 76 +++++++++++++++++++++---------------
 sound/firewire/tascam/amdtp-tascam.c | 60 +++++++++++++++++-----------
 7 files changed, 212 insertions(+), 148 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 5d2952a1a9b2..67d735e9a6a4 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -345,45 +345,59 @@ static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
 	}
 }
 
-static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	struct amdtp_am824 *p = s->protocol;
 	unsigned int pcm_frames = 0;
+	int i;
 
-	if (pcm) {
-		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			      pcm_frames);
-		pcm_frames = desc->data_blocks * p->frame_multiplier;
-	} else {
-		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
-	}
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
 
-	if (p->midi_ports) {
-		write_midi_messages(s, desc->ctx_payload, desc->data_blocks,
-				    desc->data_block_counter);
+		if (pcm) {
+			write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks * p->frame_multiplier;
+		} else {
+			write_pcm_silence(s, buf, data_blocks);
+		}
+
+		if (p->midi_ports) {
+			write_midi_messages(s, buf, data_blocks,
+					    desc->data_block_counter);
+		}
 	}
 
 	return pcm_frames;
 }
 
-static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	struct amdtp_am824 *p = s->protocol;
 	unsigned int pcm_frames = 0;
+	int i;
 
-	if (pcm) {
-		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			     pcm_frames);
-		pcm_frames = desc->data_blocks * p->frame_multiplier;
-	}
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
+
+		if (pcm) {
+			read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks * p->frame_multiplier;
+		}
 
-	if (p->midi_ports) {
-		read_midi_messages(s, desc->ctx_payload, desc->data_blocks,
-				   desc->data_block_counter);
+		if (p->midi_ports) {
+			read_midi_messages(s, buf, data_blocks,
+					   desc->data_block_counter);
+		}
 	}
 
 	return pcm_frames;
@@ -400,15 +414,14 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
 		     enum amdtp_stream_direction dir, enum cip_flags flags)
 {
-	amdtp_stream_process_data_blocks_t process_data_blocks;
+	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 
 	if (dir == AMDTP_IN_STREAM)
-		process_data_blocks = process_tx_data_blocks;
+		process_ctx_payloads = process_ir_ctx_payloads;
 	else
-		process_data_blocks = process_rx_data_blocks;
+		process_ctx_payloads = process_it_ctx_payloads;
 
 	return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
-				 process_data_blocks,
-				 sizeof(struct amdtp_am824));
+			process_ctx_payloads, sizeof(struct amdtp_am824));
 }
 EXPORT_SYMBOL_GPL(amdtp_am824_init);
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index db2feb68105c..369e75e33120 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -74,16 +74,16 @@ static void pcm_period_tasklet(unsigned long data);
  * @dir: the direction of stream
  * @flags: the packet transmission method to use
  * @fmt: the value of fmt field in CIP header
- * @process_data_blocks: callback handler to process data blocks
+ * @process_ctx_payloads: callback handler to process payloads of isoc context
  * @protocol_size: the size to allocate newly for protocol
  */
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 		      enum amdtp_stream_direction dir, enum cip_flags flags,
 		      unsigned int fmt,
-		      amdtp_stream_process_data_blocks_t process_data_blocks,
+		      amdtp_stream_process_ctx_payloads_t process_ctx_payloads,
 		      unsigned int protocol_size)
 {
-	if (process_data_blocks == NULL)
+	if (process_ctx_payloads == NULL)
 		return -EINVAL;
 
 	s->protocol = kzalloc(protocol_size, GFP_KERNEL);
@@ -102,7 +102,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->callbacked = false;
 
 	s->fmt = fmt;
-	s->process_data_blocks = process_data_blocks;
+	s->process_ctx_payloads = process_ctx_payloads;
 
 	if (dir == AMDTP_OUT_STREAM)
 		s->ctx_data.rx.syt_override = -1;
@@ -764,18 +764,13 @@ static void process_ctx_payloads(struct amdtp_stream *s,
 				 const struct pkt_desc *descs,
 				 unsigned int packets)
 {
-	int i;
-
-	for (i = 0; i < packets; ++i) {
-		const struct pkt_desc *desc = descs + i;
-		struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
-		unsigned int pcm_frames;
-
-		pcm_frames = s->process_data_blocks(s, desc, pcm);
+	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
 
-		if (pcm && pcm_frames > 0)
-			update_pcm_pointers(s, pcm, pcm_frames);
-	}
+	pcm = READ_ONCE(s->pcm);
+	pcm_frames = s->process_ctx_payloads(s, descs, packets, pcm);
+	if (pcm)
+		update_pcm_pointers(s, pcm, pcm_frames);
 }
 
 static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 0e5b85100a04..50041fa884d9 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -103,9 +103,10 @@ struct pkt_desc {
 };
 
 struct amdtp_stream;
-typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
+typedef unsigned int (*amdtp_stream_process_ctx_payloads_t)(
 						struct amdtp_stream *s,
 						const struct pkt_desc *desc,
+						unsigned int packets,
 						struct snd_pcm_substream *pcm);
 struct amdtp_stream {
 	struct fw_unit *unit;
@@ -168,13 +169,13 @@ struct amdtp_stream {
 
 	/* For backends to process data blocks. */
 	void *protocol;
-	amdtp_stream_process_data_blocks_t process_data_blocks;
+	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 };
 
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 		      enum amdtp_stream_direction dir, enum cip_flags flags,
 		      unsigned int fmt,
-		      amdtp_stream_process_data_blocks_t process_data_blocks,
+		      amdtp_stream_process_ctx_payloads_t process_ctx_payloads,
 		      unsigned int protocol_size);
 void amdtp_stream_destroy(struct amdtp_stream *s);
 
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
index cef5caf97236..d613642a2ce3 100644
--- a/sound/firewire/digi00x/amdtp-dot.c
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -341,39 +341,53 @@ void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
 		WRITE_ONCE(p->midi[port], midi);
 }
 
-static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	unsigned int pcm_frames = 0;
+	int i;
 
-	if (pcm) {
-		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			     pcm_frames);
-		pcm_frames = desc->data_blocks;
-	}
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
 
-	read_midi_messages(s, desc->ctx_payload, desc->data_blocks);
+		if (pcm) {
+			read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		}
+
+		read_midi_messages(s, buf, data_blocks);
+	}
 
 	return pcm_frames;
 }
 
-static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	unsigned int pcm_frames = 0;
+	int i;
 
-	if (pcm) {
-		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			      pcm_frames);
-		pcm_frames = desc->data_blocks;
-	} else {
-		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
-	}
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
 
-	write_midi_messages(s, desc->ctx_payload, desc->data_blocks,
-			    desc->data_block_counter);
+		if (pcm) {
+			write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		} else {
+			write_pcm_silence(s, buf, data_blocks);
+		}
+
+		write_midi_messages(s, buf, data_blocks,
+				    desc->data_block_counter);
+	}
 
 	return pcm_frames;
 }
@@ -381,20 +395,20 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
 		 enum amdtp_stream_direction dir)
 {
-	amdtp_stream_process_data_blocks_t process_data_blocks;
+	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 	enum cip_flags flags;
 
-	/* Use different mode between incoming/outgoing. */
+	// Use different mode between incoming/outgoing.
 	if (dir == AMDTP_IN_STREAM) {
 		flags = CIP_NONBLOCKING;
-		process_data_blocks = process_tx_data_blocks;
+		process_ctx_payloads = process_ir_ctx_payloads;
 	} else {
 		flags = CIP_BLOCKING;
-		process_data_blocks = process_rx_data_blocks;
+		process_ctx_payloads = process_it_ctx_payloads;
 	}
 
 	return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
-				 process_data_blocks, sizeof(struct amdtp_dot));
+				process_ctx_payloads, sizeof(struct amdtp_dot));
 }
 
 void amdtp_dot_reset(struct amdtp_stream *s)
diff --git a/sound/firewire/fireface/amdtp-ff.c b/sound/firewire/fireface/amdtp-ff.c
index cb2073cf146b..119c0076b17a 100644
--- a/sound/firewire/fireface/amdtp-ff.c
+++ b/sound/firewire/fireface/amdtp-ff.c
@@ -112,34 +112,47 @@ int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s,
 	return amdtp_stream_add_pcm_hw_constraints(s, runtime);
 }
 
-static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
+static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
+					   const struct pkt_desc *descs,
+					   unsigned int packets,
 					   struct snd_pcm_substream *pcm)
 {
 	unsigned int pcm_frames = 0;
-
-	if (pcm) {
-		write_pcm_s32(s, pcm, (__le32 *)desc->ctx_payload,
-			      desc->data_blocks, pcm_frames);
-		pcm_frames = desc->data_blocks;
-	} else {
-		write_pcm_silence(s, (__le32 *)desc->ctx_payload,
-				  desc->data_blocks);
+	int i;
+
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__le32 *buf = (__le32 *)desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
+
+		if (pcm) {
+			write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		} else {
+			write_pcm_silence(s, buf, data_blocks);
+		}
 	}
 
 	return pcm_frames;
 }
 
-static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	unsigned int pcm_frames = 0;
+	int i;
 
-	if (pcm) {
-		read_pcm_s32(s, pcm, (__le32 *)desc->ctx_payload,
-			     desc->data_blocks, pcm_frames);
-		pcm_frames = desc->data_blocks;
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__le32 *buf = (__le32 *)desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
+
+		if (pcm) {
+			read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		}
 	}
 
 	return pcm_frames;
@@ -148,13 +161,13 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 int amdtp_ff_init(struct amdtp_stream *s, struct fw_unit *unit,
 		  enum amdtp_stream_direction dir)
 {
-	amdtp_stream_process_data_blocks_t process_data_blocks;
+	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 
 	if (dir == AMDTP_IN_STREAM)
-		process_data_blocks = process_tx_data_blocks;
+		process_ctx_payloads = process_ir_ctx_payloads;
 	else
-		process_data_blocks = process_rx_data_blocks;
+		process_ctx_payloads = process_it_ctx_payloads;
 
 	return amdtp_stream_init(s, unit, dir, CIP_NO_HEADER, 0,
-				 process_data_blocks, sizeof(struct amdtp_ff));
+				 process_ctx_payloads, sizeof(struct amdtp_ff));
 }
diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 21c6bfdf47c0..05a6e218b0ad 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -310,23 +310,30 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 	}
 }
 
-static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	struct amdtp_motu *p = s->protocol;
 	unsigned int pcm_frames = 0;
+	int i;
+
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
 
-	trace_data_block_sph(s, desc->data_blocks, desc->ctx_payload);
-	trace_data_block_message(s, desc->data_blocks, desc->ctx_payload);
+		trace_data_block_sph(s, data_blocks, buf);
+		trace_data_block_message(s, data_blocks, buf);
 
-	if (p->midi_ports)
-		read_midi_messages(s, desc->ctx_payload, desc->data_blocks);
+		if (p->midi_ports)
+			read_midi_messages(s, buf, data_blocks);
 
-	if (pcm) {
-		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			     pcm_frames);
-		pcm_frames = desc->data_blocks;
+		if (pcm) {
+			read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		}
 	}
 
 	return pcm_frames;
@@ -374,30 +381,37 @@ static void write_sph(struct amdtp_stream *s, __be32 *buffer,
 	}
 }
 
-static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	struct amdtp_motu *p = s->protocol;
 	unsigned int pcm_frames = 0;
+	int i;
 
-	/* TODO: how to interact control messages between userspace? */
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
 
-	if (p->midi_ports)
-		write_midi_messages(s, desc->ctx_payload, desc->data_blocks);
+		// TODO: how to interact control messages between userspace?
 
-	if (pcm) {
-		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			      pcm_frames);
-		pcm_frames = desc->data_blocks;
-	} else {
-		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
-	}
+		if (p->midi_ports)
+			write_midi_messages(s, buf, data_blocks);
 
-	write_sph(s, desc->ctx_payload, desc->data_blocks);
+		if (pcm) {
+			write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		} else {
+			write_pcm_silence(s, buf, data_blocks);
+		}
 
-	trace_data_block_sph(s, desc->data_blocks, desc->ctx_payload);
-	trace_data_block_message(s, desc->data_blocks, desc->ctx_payload);
+		write_sph(s, buf, data_blocks);
+
+		trace_data_block_sph(s, data_blocks, buf);
+		trace_data_block_message(s, data_blocks, buf);
+	}
 
 	return pcm_frames;
 }
@@ -406,13 +420,13 @@ int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
 		    enum amdtp_stream_direction dir,
 		    const struct snd_motu_protocol *const protocol)
 {
-	amdtp_stream_process_data_blocks_t process_data_blocks;
+	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 	int fmt = CIP_FMT_MOTU;
 	int flags = CIP_BLOCKING;
 	int err;
 
 	if (dir == AMDTP_IN_STREAM) {
-		process_data_blocks = process_tx_data_blocks;
+		process_ctx_payloads = process_ir_ctx_payloads;
 
 		/*
 		 * Units of version 3 transmits packets with invalid CIP header
@@ -431,11 +445,11 @@ int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
 				 CIP_SKIP_DBC_ZERO_CHECK;
 		}
 	} else {
-		process_data_blocks = process_rx_data_blocks;
+		process_ctx_payloads = process_it_ctx_payloads;
 		flags |= CIP_DBC_IS_END_EVENT;
 	}
 
-	err = amdtp_stream_init(s, unit, dir, flags, fmt, process_data_blocks,
+	err = amdtp_stream_init(s, unit, dir, flags, fmt, process_ctx_payloads,
 				sizeof(struct amdtp_motu));
 	if (err < 0)
 		return err;
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index 3c9abf422f24..e80bb84c43f6 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -175,35 +175,49 @@ static void read_status_messages(struct amdtp_stream *s,
 	}
 }
 
-static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	unsigned int pcm_frames = 0;
+	int i;
 
-	if (pcm) {
-		read_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			     pcm_frames);
-		pcm_frames += desc->data_blocks;
-	}
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
 
-	read_status_messages(s, desc->ctx_payload, desc->data_blocks);
+		if (pcm) {
+			read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		}
+
+		read_status_messages(s, buf, data_blocks);
+	}
 
 	return pcm_frames;
 }
 
-static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
-					   const struct pkt_desc *desc,
-					   struct snd_pcm_substream *pcm)
+static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
+					    const struct pkt_desc *descs,
+					    unsigned int packets,
+					    struct snd_pcm_substream *pcm)
 {
 	unsigned int pcm_frames = 0;
+	int i;
 
-	if (pcm) {
-		write_pcm_s32(s, pcm, desc->ctx_payload, desc->data_blocks,
-			      pcm_frames);
-		pcm_frames += desc->data_blocks;
-	} else {
-		write_pcm_silence(s, desc->ctx_payload, desc->data_blocks);
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
+
+		if (pcm) {
+			write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
+			pcm_frames += data_blocks;
+		} else {
+			write_pcm_silence(s, buf, data_blocks);
+		}
 	}
 
 	return pcm_frames;
@@ -212,22 +226,22 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
 		    enum amdtp_stream_direction dir, unsigned int pcm_channels)
 {
-	amdtp_stream_process_data_blocks_t process_data_blocks;
+	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 	struct amdtp_tscm *p;
 	unsigned int fmt;
 	int err;
 
 	if (dir == AMDTP_IN_STREAM) {
 		fmt = AMDTP_FMT_TSCM_TX;
-		process_data_blocks = process_tx_data_blocks;
+		process_ctx_payloads = process_ir_ctx_payloads;
 	} else {
 		fmt = AMDTP_FMT_TSCM_RX;
-		process_data_blocks = process_rx_data_blocks;
+		process_ctx_payloads = process_it_ctx_payloads;
 	}
 
 	err = amdtp_stream_init(s, unit, dir,
-				CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt,
-				process_data_blocks, sizeof(struct amdtp_tscm));
+			CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt,
+			process_ctx_payloads, sizeof(struct amdtp_tscm));
 	if (err < 0)
 		return 0;
 
-- 
cgit v1.2.3-59-g8ed1b


From d2d5a6b0253d30ef04d513447cdd00df4fa627c3 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 Jul 2019 12:37:10 +0900
Subject: ALSA: firewire-motu: more code refactoring for MOTU data block
 processing layer

MOTU data block processing layer has some tracepoints events. This
commit is code refactoring to split probing the events from processing
data blocks.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/motu/amdtp-motu.c | 50 +++++++++++++++++++++++++++++-----------
 1 file changed, 36 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 05a6e218b0ad..0fd36e469ad0 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -310,6 +310,22 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 	}
 }
 
+static void probe_tracepoints_events(struct amdtp_stream *s,
+				     const struct pkt_desc *descs,
+				     unsigned int packets)
+{
+	int i;
+
+	for (i = 0; i < packets; ++i) {
+		const struct pkt_desc *desc = descs + i;
+		__be32 *buf = desc->ctx_payload;
+		unsigned int data_blocks = desc->data_blocks;
+
+		trace_data_block_sph(s, data_blocks, buf);
+		trace_data_block_message(s, data_blocks, buf);
+	}
+}
+
 static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s,
 					    const struct pkt_desc *descs,
 					    unsigned int packets,
@@ -319,23 +335,26 @@ static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s,
 	unsigned int pcm_frames = 0;
 	int i;
 
+	// For data block processing.
 	for (i = 0; i < packets; ++i) {
 		const struct pkt_desc *desc = descs + i;
 		__be32 *buf = desc->ctx_payload;
 		unsigned int data_blocks = desc->data_blocks;
 
-		trace_data_block_sph(s, data_blocks, buf);
-		trace_data_block_message(s, data_blocks, buf);
-
-		if (p->midi_ports)
-			read_midi_messages(s, buf, data_blocks);
-
 		if (pcm) {
 			read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
 			pcm_frames += data_blocks;
 		}
+
+		if (p->midi_ports)
+			read_midi_messages(s, buf, data_blocks);
 	}
 
+	// For tracepoints.
+	if (trace_data_block_sph_enabled() ||
+	    trace_data_block_message_enabled())
+		probe_tracepoints_events(s, descs, packets);
+
 	return pcm_frames;
 }
 
@@ -390,16 +409,12 @@ static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
 	unsigned int pcm_frames = 0;
 	int i;
 
+	// For data block processing.
 	for (i = 0; i < packets; ++i) {
 		const struct pkt_desc *desc = descs + i;
 		__be32 *buf = desc->ctx_payload;
 		unsigned int data_blocks = desc->data_blocks;
 
-		// TODO: how to interact control messages between userspace?
-
-		if (p->midi_ports)
-			write_midi_messages(s, buf, data_blocks);
-
 		if (pcm) {
 			write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames);
 			pcm_frames += data_blocks;
@@ -407,12 +422,19 @@ static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
 			write_pcm_silence(s, buf, data_blocks);
 		}
 
-		write_sph(s, buf, data_blocks);
+		if (p->midi_ports)
+			write_midi_messages(s, buf, data_blocks);
 
-		trace_data_block_sph(s, data_blocks, buf);
-		trace_data_block_message(s, data_blocks, buf);
+		// TODO: how to interact control messages between userspace?
+
+		write_sph(s, buf, data_blocks);
 	}
 
+	// For tracepoints.
+	if (trace_data_block_sph_enabled() ||
+	    trace_data_block_message_enabled())
+		probe_tracepoints_events(s, descs, packets);
+
 	return pcm_frames;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 748fd07e2b9ca4132e3d2aae25395aedc4d1aee8 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Mon, 22 Jul 2019 14:52:09 +0100
Subject: ASoC: madera: Read device tree configuration

Read the configuration of the Madera ASoC driver from device tree.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190722135209.30302-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/madera.c | 97 +++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 97 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c
index 1b1be19a2f99..5f1e32a5a855 100644
--- a/sound/soc/codecs/madera.c
+++ b/sound/soc/codecs/madera.c
@@ -300,6 +300,100 @@ int madera_free_overheat(struct madera_priv *priv)
 }
 EXPORT_SYMBOL_GPL(madera_free_overheat);
 
+static int madera_get_variable_u32_array(struct device *dev,
+					 const char *propname,
+					 u32 *dest, int n_max,
+					 int multiple)
+{
+	int n, ret;
+
+	n = device_property_count_u32(dev, propname);
+	if (n < 0) {
+		if (n == -EINVAL)
+			return 0;	/* missing, ignore */
+
+		dev_warn(dev, "%s malformed (%d)\n", propname, n);
+
+		return n;
+	} else if ((n % multiple) != 0) {
+		dev_warn(dev, "%s not a multiple of %d entries\n",
+			 propname, multiple);
+
+		return -EINVAL;
+	}
+
+	if (n > n_max)
+		n = n_max;
+
+	ret = device_property_read_u32_array(dev, propname, dest, n);
+	if (ret < 0)
+		return ret;
+
+	return n;
+}
+
+static void madera_prop_get_inmode(struct madera_priv *priv)
+{
+	struct madera *madera = priv->madera;
+	struct madera_codec_pdata *pdata = &madera->pdata.codec;
+	u32 tmp[MADERA_MAX_INPUT * MADERA_MAX_MUXED_CHANNELS];
+	int n, i, in_idx, ch_idx;
+
+	BUILD_BUG_ON(ARRAY_SIZE(pdata->inmode) != MADERA_MAX_INPUT);
+	BUILD_BUG_ON(ARRAY_SIZE(pdata->inmode[0]) != MADERA_MAX_MUXED_CHANNELS);
+
+	n = madera_get_variable_u32_array(madera->dev, "cirrus,inmode",
+					  tmp, ARRAY_SIZE(tmp),
+					  MADERA_MAX_MUXED_CHANNELS);
+	if (n < 0)
+		return;
+
+	in_idx = 0;
+	ch_idx = 0;
+	for (i = 0; i < n; ++i) {
+		pdata->inmode[in_idx][ch_idx] = tmp[i];
+
+		if (++ch_idx == MADERA_MAX_MUXED_CHANNELS) {
+			ch_idx = 0;
+			++in_idx;
+		}
+	}
+}
+
+static void madera_prop_get_pdata(struct madera_priv *priv)
+{
+	struct madera *madera = priv->madera;
+	struct madera_codec_pdata *pdata = &madera->pdata.codec;
+	u32 out_mono[ARRAY_SIZE(pdata->out_mono)];
+	int i, n;
+
+	madera_prop_get_inmode(priv);
+
+	n = madera_get_variable_u32_array(madera->dev, "cirrus,out-mono",
+					  out_mono, ARRAY_SIZE(out_mono), 1);
+	if (n > 0)
+		for (i = 0; i < n; ++i)
+			pdata->out_mono[i] = !!out_mono[i];
+
+	madera_get_variable_u32_array(madera->dev,
+				      "cirrus,max-channels-clocked",
+				      pdata->max_channels_clocked,
+				      ARRAY_SIZE(pdata->max_channels_clocked),
+				      1);
+
+	madera_get_variable_u32_array(madera->dev, "cirrus,pdm-fmt",
+				      pdata->pdm_fmt,
+				      ARRAY_SIZE(pdata->pdm_fmt), 1);
+
+	madera_get_variable_u32_array(madera->dev, "cirrus,pdm-mute",
+				      pdata->pdm_mute,
+				      ARRAY_SIZE(pdata->pdm_mute), 1);
+
+	madera_get_variable_u32_array(madera->dev, "cirrus,dmic-ref",
+				      pdata->dmic_ref,
+				      ARRAY_SIZE(pdata->dmic_ref), 1);
+}
+
 int madera_core_init(struct madera_priv *priv)
 {
 	int i;
@@ -308,6 +402,9 @@ int madera_core_init(struct madera_priv *priv)
 	BUILD_BUG_ON(!madera_mixer_texts[MADERA_NUM_MIXER_INPUTS - 1]);
 	BUILD_BUG_ON(!madera_mixer_values[MADERA_NUM_MIXER_INPUTS - 1]);
 
+	if (!dev_get_platdata(priv->madera->dev))
+		madera_prop_get_pdata(priv);
+
 	mutex_init(&priv->rate_lock);
 
 	for (i = 0; i < MADERA_MAX_HP_OUTPUT; i++)
-- 
cgit v1.2.3-59-g8ed1b


From f8f85216f8d309daadb37aba8a4b0826783d8747 Mon Sep 17 00:00:00 2001
From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Date: Mon, 22 Jul 2019 23:15:28 +0200
Subject: ASoC: pcm3168a: Fix a typo in the name of a constant

There is a typo in PCM1368A_MAX_SYSCLK, it should be PCM3168A_MAX_SYSCLK
(1 and 3 switched in 3168)

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20190722211528.26600-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/pcm3168a.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index f1104d7d6426..5d59ce254821 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -263,7 +263,7 @@ static unsigned int pcm3168a_scki_ratios[] = {
 #define PCM3168A_NUM_SCKI_RATIOS_DAC	ARRAY_SIZE(pcm3168a_scki_ratios)
 #define PCM3168A_NUM_SCKI_RATIOS_ADC	(ARRAY_SIZE(pcm3168a_scki_ratios) - 2)
 
-#define PCM1368A_MAX_SYSCLK		36864000
+#define PCM3168A_MAX_SYSCLK		36864000
 
 static int pcm3168a_reset(struct pcm3168a_priv *pcm3168a)
 {
@@ -296,7 +296,7 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai,
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component);
 	int ret;
 
-	if (freq > PCM1368A_MAX_SYSCLK)
+	if (freq > PCM3168A_MAX_SYSCLK)
 		return -EINVAL;
 
 	ret = clk_set_rate(pcm3168a->scki, freq);
-- 
cgit v1.2.3-59-g8ed1b


From f1b1b9b136827915624136624ff54aba5890a15b Mon Sep 17 00:00:00 2001
From: Pan Xiuli <xiuli.pan@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:42 -0500
Subject: ASoC: SOF: pci: mark last_busy value at runtime PM init

If last_busy value is not set at runtime PM enable, the device will be
suspend immediately after usage counter is 0. Set the last_busy value to
make sure delay is working at first boot up.

Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/sof-pci-dev.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index 65d1bac4c6b8..6fd3df7c57a3 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -223,6 +223,9 @@ static void sof_pci_probe_complete(struct device *dev)
 	 */
 	pm_runtime_allow(dev);
 
+	/* mark last_busy for pm_runtime to make sure not suspend immediately */
+	pm_runtime_mark_last_busy(dev);
+
 	/* follow recommendation in pci-driver.c to decrement usage counter */
 	pm_runtime_put_noidle(dev);
 }
-- 
cgit v1.2.3-59-g8ed1b


From 04c8027764bc82a325d3abc6f39a6a4642a937cb Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:43 -0500
Subject: ASoC: SOF: reset DMA state in prepare

When application goes through SUSPEND/STOP->PREPARE->START
cycle, we should always reprogram the SOF device to start
DMA from a known state so that hw_ptr/appl_ptrs remain valid.
This is expected by ALSA core as it resets the buffer
state as part of prepare (see snd_pcm_do_prepare()).

Fix the issue by forcing reconfiguration of the FW with
STREAM_PCM_PARAMS in prepare(). Use combined logic to handle
prepare and the existing flow to reprogram hw-params after
system suspend.

Without the fix, first call to pcm pointer() will return
an invalid hw_ptr and application may immediately observe XRUN
status, unless "start_threshold" SW parameter is set to maximum
value by the application.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/pcm.c      | 27 +++++++++++++++------------
 sound/soc/sof/pm.c       |  2 +-
 sound/soc/sof/sof-priv.h |  2 +-
 3 files changed, 17 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 334e9d59b1ba..3b8955e755b2 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -208,12 +208,11 @@ static int sof_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
+	spcm->prepared[substream->stream] = true;
+
 	/* save pcm hw_params */
 	memcpy(&spcm->params[substream->stream], params, sizeof(*params));
 
-	/* clear hw_params_upon_resume flag */
-	spcm->hw_params_upon_resume[substream->stream] = 0;
-
 	return ret;
 }
 
@@ -236,6 +235,9 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (!spcm)
 		return -EINVAL;
 
+	if (!spcm->prepared[substream->stream])
+		return 0;
+
 	dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id,
 		substream->stream);
 
@@ -258,6 +260,8 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (ret < 0)
 		dev_err(sdev->dev, "error: platform hw free failed\n");
 
+	spcm->prepared[substream->stream] = false;
+
 	return ret;
 }
 
@@ -278,11 +282,7 @@ static int sof_pcm_prepare(struct snd_pcm_substream *substream)
 	if (!spcm)
 		return -EINVAL;
 
-	/*
-	 * check if hw_params needs to be set-up again.
-	 * This is only needed when resuming from system sleep.
-	 */
-	if (!spcm->hw_params_upon_resume[substream->stream])
+	if (spcm->prepared[substream->stream])
 		return 0;
 
 	dev_dbg(sdev->dev, "pcm: prepare stream %d dir %d\n", spcm->pcm.pcm_id,
@@ -311,6 +311,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_sof_pcm *spcm;
 	struct sof_ipc_stream stream;
 	struct sof_ipc_reply reply;
+	bool reset_hw_params = false;
 	int ret;
 
 	/* nothing to do for BE */
@@ -351,6 +352,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
+		reset_hw_params = true;
 		break;
 	default:
 		dev_err(sdev->dev, "error: unhandled trigger cmd %d\n", cmd);
@@ -363,17 +365,17 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
 				 sizeof(stream), &reply, sizeof(reply));
 
-	if (ret < 0 || cmd != SNDRV_PCM_TRIGGER_SUSPEND)
+	if (ret < 0 || !reset_hw_params)
 		return ret;
 
 	/*
-	 * The hw_free op is usually called when the pcm stream is closed.
-	 * Since the stream is not closed during suspend, the DSP needs to be
-	 * notified explicitly to free pcm to prevent errors upon resume.
+	 * In case of stream is stopped, DSP must be reprogrammed upon
+	 * restart, so free PCM here.
 	 */
 	stream.hdr.size = sizeof(stream);
 	stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE;
 	stream.comp_id = spcm->stream[substream->stream].comp_id;
+	spcm->prepared[substream->stream] = false;
 
 	/* send IPC to the DSP */
 	return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
@@ -481,6 +483,7 @@ static int sof_pcm_open(struct snd_pcm_substream *substream)
 	spcm->stream[substream->stream].posn.host_posn = 0;
 	spcm->stream[substream->stream].posn.dai_posn = 0;
 	spcm->stream[substream->stream].substream = substream;
+	spcm->prepared[substream->stream] = false;
 
 	ret = snd_sof_pcm_platform_open(sdev, substream);
 	if (ret < 0)
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 278abfd10490..48c6d78d72e2 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -233,7 +233,7 @@ static int sof_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 
 			state = substream->runtime->status->state;
 			if (state == SNDRV_PCM_STATE_SUSPENDED)
-				spcm->hw_params_upon_resume[dir] = 1;
+				spcm->prepared[dir] = false;
 		}
 	}
 
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index b8c0b2a22684..fa5cb7d2a660 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -297,7 +297,7 @@ struct snd_sof_pcm {
 	struct snd_sof_pcm_stream stream[2];
 	struct list_head list;	/* list in sdev pcm list */
 	struct snd_pcm_hw_params params[2];
-	int hw_params_upon_resume[2]; /* set up hw_params upon resume */
+	bool prepared[2]; /* PCM_PARAMS set successfully */
 };
 
 /* ALSA SOF Kcontrol device */
-- 
cgit v1.2.3-59-g8ed1b


From a49b687192752bd373f33551520cc98389460ad0 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:44 -0500
Subject: ASoC: SOF: use common code to send PCM_FREE IPC

Remove duplicated code by using a common helper function
to send the PCM_FREE IPC message to FW.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/pcm.c | 50 +++++++++++++++++++++++++-------------------------
 1 file changed, 25 insertions(+), 25 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 3b8955e755b2..8612896673a5 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -216,6 +216,27 @@ static int sof_pcm_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream,
+				struct snd_sof_dev *sdev,
+				struct snd_sof_pcm *spcm)
+{
+	struct sof_ipc_stream stream;
+	struct sof_ipc_reply reply;
+	int ret;
+
+	stream.hdr.size = sizeof(stream);
+	stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE;
+	stream.comp_id = spcm->stream[substream->stream].comp_id;
+
+	/* send IPC to the DSP */
+	ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
+				 sizeof(stream), &reply, sizeof(reply));
+	if (!ret)
+		spcm->prepared[substream->stream] = false;
+
+	return ret;
+}
+
 static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -223,8 +244,6 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
 	struct snd_sof_pcm *spcm;
-	struct sof_ipc_stream stream;
-	struct sof_ipc_reply reply;
 	int ret;
 
 	/* nothing to do for BE */
@@ -241,13 +260,7 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id,
 		substream->stream);
 
-	stream.hdr.size = sizeof(stream);
-	stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE;
-	stream.comp_id = spcm->stream[substream->stream].comp_id;
-
-	/* send IPC to the DSP */
-	ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
-				 sizeof(stream), &reply, sizeof(reply));
+	ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
 
 	snd_pcm_lib_free_pages(substream);
 
@@ -260,8 +273,6 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (ret < 0)
 		dev_err(sdev->dev, "error: platform hw free failed\n");
 
-	spcm->prepared[substream->stream] = false;
-
 	return ret;
 }
 
@@ -365,21 +376,10 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
 				 sizeof(stream), &reply, sizeof(reply));
 
-	if (ret < 0 || !reset_hw_params)
-		return ret;
-
-	/*
-	 * In case of stream is stopped, DSP must be reprogrammed upon
-	 * restart, so free PCM here.
-	 */
-	stream.hdr.size = sizeof(stream);
-	stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE;
-	stream.comp_id = spcm->stream[substream->stream].comp_id;
-	spcm->prepared[substream->stream] = false;
+	if (!ret && reset_hw_params)
+		ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
 
-	/* send IPC to the DSP */
-	return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
-				  sizeof(stream), &reply, sizeof(reply));
+	return ret;
 }
 
 static snd_pcm_uframes_t sof_pcm_pointer(struct snd_pcm_substream *substream)
-- 
cgit v1.2.3-59-g8ed1b


From 6dd78b310b8d4ab5d0b851e369b5b3d46c6a32a9 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:45 -0500
Subject: ASoC: SOF: ipc: use timeout configured at probe

Do not hardcode IPC timeout value in ipc.c, but rather use the timeout
value configured during device probe. For platforms that do not override
the IPC timeout, default value TIMEOUT_DEFAULT_IPC_MS has already been
defined in core.c.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/ipc.c | 8 +-------
 1 file changed, 1 insertion(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c
index 20dfca9c93b7..b2f359d2f7e5 100644
--- a/sound/soc/sof/ipc.c
+++ b/sound/soc/sof/ipc.c
@@ -17,12 +17,6 @@
 #include "sof-priv.h"
 #include "ops.h"
 
-/*
- * IPC message default size and timeout (ms).
- * TODO: allow platforms to set size and timeout.
- */
-#define IPC_TIMEOUT_MS		300
-
 static void ipc_trace_message(struct snd_sof_dev *sdev, u32 msg_id);
 static void ipc_stream_message(struct snd_sof_dev *sdev, u32 msg_cmd);
 
@@ -211,7 +205,7 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg,
 
 	/* wait for DSP IPC completion */
 	ret = wait_event_timeout(msg->waitq, msg->ipc_complete,
-				 msecs_to_jiffies(IPC_TIMEOUT_MS));
+				 msecs_to_jiffies(sdev->ipc_timeout));
 
 	if (ret == 0) {
 		dev_err(sdev->dev, "error: ipc timed out for 0x%x size %d\n",
-- 
cgit v1.2.3-59-g8ed1b


From c9b54a3360166ca2fec11d55c86b5e685e56bab8 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:46 -0500
Subject: ASoC: SOF: core: increase default IPC timeouts

Increase the default timeout values for boot (100ms to 2sec) and
IPC message sending (5ms to 500ms). The values should be overridden
with values from platform data.

There is no functional need to have such short timeouts as both boot
and IPC send errors are considered fatal errors. More relaxed timeouts
are convenient when running the driver on top of emulation such as QEMU.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/core.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 5beda47cdf9f..81f28f7ff1a0 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -17,8 +17,8 @@
 #include "ops.h"
 
 /* SOF defaults if not provided by the platform in ms */
-#define TIMEOUT_DEFAULT_IPC_MS  5
-#define TIMEOUT_DEFAULT_BOOT_MS 100
+#define TIMEOUT_DEFAULT_IPC_MS  500
+#define TIMEOUT_DEFAULT_BOOT_MS 2000
 
 /*
  * Generic object lookup APIs.
-- 
cgit v1.2.3-59-g8ed1b


From ce8234a6357ec8f81b977d8536ff941dbd4162f3 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Mon, 22 Jul 2019 09:13:47 -0500
Subject: ASoC: SOF: Introduce snd_sof_dsp_get_bar_index ops

FW encapsulates information about section types (e.g DRAM, IRAM)
inside module block header. This information can be used in order
to correctly load the section to the appropriate place in memory.

SOF Linux driver needs to know for each platform how to map the
section type with the corresponding memory BAR. So, this patch
introduces get_bar_index, a new operation inside snd_sof_dsp_ops.

Intel platforms, usually load all the section in a contiguous memory
area (usually denoted by sdev->mmio_bar) so things are relatively
simple there. Anyhow, on i.MX8 IRAM and DRAM for example are mapped
to distinct BARs.

By default, if no get_bar function is provided the core implementation
will always return sdev->mmio_bar so that there will be no need for
a change to existing Intel code.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/ops.h      | 19 +++++++++++++++++++
 sound/soc/sof/sof-priv.h |  3 +++
 2 files changed, 22 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index b1c27615b805..8fda42748f39 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -100,6 +100,25 @@ static inline int snd_sof_dsp_post_fw_run(struct snd_sof_dev *sdev)
 	return 0;
 }
 
+/* misc */
+
+/**
+ * snd_sof_dsp_get_bar_index - Maps a section type with a BAR index
+ *
+ * @sdev: sof device
+ * @type: section type as described by snd_sof_fw_blk_type
+ *
+ * Returns the corresponding BAR index (a positive integer) or -EINVAL
+ * in case there is no mapping
+ */
+static inline int snd_sof_dsp_get_bar_index(struct snd_sof_dev *sdev, u32 type)
+{
+	if (sof_ops(sdev)->get_bar_index)
+		return sof_ops(sdev)->get_bar_index(sdev, type);
+
+	return sdev->mmio_bar;
+}
+
 /* power management */
 static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev)
 {
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index fa5cb7d2a660..a6ec9115a9fb 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -196,6 +196,9 @@ struct snd_sof_dsp_ops {
 	int (*trace_trigger)(struct snd_sof_dev *sdev,
 			     int cmd); /* optional */
 
+	/* misc */
+	int (*get_bar_index)(struct snd_sof_dev *sdev,
+			     u32 type); /* optional */
 	/* DAI ops */
 	struct snd_soc_dai_driver *drv;
 	int num_drv;
-- 
cgit v1.2.3-59-g8ed1b


From 7198879ef57775df3b1a75c62bb9e313c56876d4 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Mon, 22 Jul 2019 09:13:48 -0500
Subject: ASoC: SOF: loader: Use the BAR provided by FW

Make sure to use the newly introduced function snd_sof_dsp_get_bar_index
that converts the section type to appropriate BAR index.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/loader.c | 11 +++++++++--
 1 file changed, 9 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 952a19091c58..e75da8aa7d4a 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -92,7 +92,7 @@ int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev,
 				struct snd_sof_mod_hdr *module)
 {
 	struct snd_sof_blk_hdr *block;
-	int count;
+	int count, bar;
 	u32 offset;
 	size_t remaining;
 
@@ -128,6 +128,13 @@ int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev,
 		case SOF_FW_BLK_TYPE_IRAM:
 		case SOF_FW_BLK_TYPE_DRAM:
 			offset = block->offset;
+			bar = snd_sof_dsp_get_bar_index(sdev, block->type);
+			if (bar < 0) {
+				dev_err(sdev->dev,
+					"error: no BAR mapping for block type 0x%x\n",
+					block->type);
+				return bar;
+			}
 			break;
 		default:
 			dev_err(sdev->dev, "error: bad type 0x%x for block 0x%x\n",
@@ -145,7 +152,7 @@ int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev,
 				block->size);
 			return -EINVAL;
 		}
-		snd_sof_dsp_block_write(sdev, sdev->mmio_bar, offset,
+		snd_sof_dsp_block_write(sdev, bar, offset,
 					block + 1, block->size);
 
 		if (remaining < block->size) {
-- 
cgit v1.2.3-59-g8ed1b


From 441c58cf8e5e0c594b3d1b489e7810ec7248820d Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Mon, 22 Jul 2019 09:13:49 -0500
Subject: ASoC: SOF: loader: Don't ignore SRAM block types

On i.MX8 data/heap/stack is kept in System RAM so
do not ignore SRAM block types received from FW.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/loader.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index e75da8aa7d4a..93cb8fd0844f 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -123,10 +123,11 @@ int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev,
 
 		switch (block->type) {
 		case SOF_FW_BLK_TYPE_RSRVD0:
-		case SOF_FW_BLK_TYPE_SRAM...SOF_FW_BLK_TYPE_RSRVD14:
+		case SOF_FW_BLK_TYPE_ROM...SOF_FW_BLK_TYPE_RSRVD14:
 			continue;	/* not handled atm */
 		case SOF_FW_BLK_TYPE_IRAM:
 		case SOF_FW_BLK_TYPE_DRAM:
+		case SOF_FW_BLK_TYPE_SRAM:
 			offset = block->offset;
 			bar = snd_sof_dsp_get_bar_index(sdev, block->type);
 			if (bar < 0) {
-- 
cgit v1.2.3-59-g8ed1b


From 1c38c9223da3af619d35f052ad6f4bbaa0f08ac2 Mon Sep 17 00:00:00 2001
From: Fred Oh <fred.oh@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:50 -0500
Subject: ASoC: SOF: remove unused state variable in suspend function

Remove unused and no plan to use variable from suspend function.

Signed-off-by: Fred Oh <fred.oh@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-dsp.c | 10 +++++-----
 sound/soc/sof/intel/hda.h     |  4 ++--
 sound/soc/sof/ops.h           |  9 ++++-----
 sound/soc/sof/pm.c            |  4 ++--
 sound/soc/sof/sof-priv.h      |  5 ++---
 5 files changed, 15 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 91de4785b6a3..4315896b3f07 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -282,7 +282,7 @@ void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev)
 			HDA_DSP_REG_HIPCCTL_BUSY | HDA_DSP_REG_HIPCCTL_DONE, 0);
 }
 
-static int hda_suspend(struct snd_sof_dev *sdev, int state)
+static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
 {
 	struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
 	const struct sof_intel_dsp_desc *chip = hda->desc;
@@ -431,19 +431,19 @@ int hda_dsp_runtime_idle(struct snd_sof_dev *sdev)
 	return 0;
 }
 
-int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev, int state)
+int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev)
 {
 	/* stop hda controller and power dsp off */
-	return hda_suspend(sdev, state);
+	return hda_suspend(sdev, true);
 }
 
-int hda_dsp_suspend(struct snd_sof_dev *sdev, int state)
+int hda_dsp_suspend(struct snd_sof_dev *sdev)
 {
 	struct hdac_bus *bus = sof_to_bus(sdev);
 	int ret;
 
 	/* stop hda controller and power dsp off */
-	ret = hda_suspend(sdev, state);
+	ret = hda_suspend(sdev, false);
 	if (ret < 0) {
 		dev_err(bus->dev, "error: suspending dsp\n");
 		return ret;
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index d9c17146200b..84d5fd021ccb 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -453,9 +453,9 @@ int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev,
 void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev);
 void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev);
 
-int hda_dsp_suspend(struct snd_sof_dev *sdev, int state);
+int hda_dsp_suspend(struct snd_sof_dev *sdev);
 int hda_dsp_resume(struct snd_sof_dev *sdev);
-int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev, int state);
+int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev);
 int hda_dsp_runtime_resume(struct snd_sof_dev *sdev);
 int hda_dsp_runtime_idle(struct snd_sof_dev *sdev);
 int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev);
diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index 8fda42748f39..793c1aea0c53 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -128,10 +128,10 @@ static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev)
 	return 0;
 }
 
-static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev, int state)
+static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev)
 {
 	if (sof_ops(sdev)->suspend)
-		return sof_ops(sdev)->suspend(sdev, state);
+		return sof_ops(sdev)->suspend(sdev);
 
 	return 0;
 }
@@ -144,11 +144,10 @@ static inline int snd_sof_dsp_runtime_resume(struct snd_sof_dev *sdev)
 	return 0;
 }
 
-static inline int snd_sof_dsp_runtime_suspend(struct snd_sof_dev *sdev,
-					      int state)
+static inline int snd_sof_dsp_runtime_suspend(struct snd_sof_dev *sdev)
 {
 	if (sof_ops(sdev)->runtime_suspend)
-		return sof_ops(sdev)->runtime_suspend(sdev, state);
+		return sof_ops(sdev)->runtime_suspend(sdev);
 
 	return 0;
 }
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 48c6d78d72e2..e23beaeefe00 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -377,9 +377,9 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 
 	/* power down all DSP cores */
 	if (runtime_suspend)
-		ret = snd_sof_dsp_runtime_suspend(sdev, 0);
+		ret = snd_sof_dsp_runtime_suspend(sdev);
 	else
-		ret = snd_sof_dsp_suspend(sdev, 0);
+		ret = snd_sof_dsp_suspend(sdev);
 	if (ret < 0)
 		dev_err(sdev->dev,
 			"error: failed to power down DSP during suspend %d\n",
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index a6ec9115a9fb..b8c9274ccf42 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -171,10 +171,9 @@ struct snd_sof_dsp_ops {
 	int (*post_fw_run)(struct snd_sof_dev *sof_dev); /* optional */
 
 	/* DSP PM */
-	int (*suspend)(struct snd_sof_dev *sof_dev, int state); /* optional */
+	int (*suspend)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*resume)(struct snd_sof_dev *sof_dev); /* optional */
-	int (*runtime_suspend)(struct snd_sof_dev *sof_dev,
-			       int state); /* optional */
+	int (*runtime_suspend)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*runtime_resume)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*runtime_idle)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*set_hw_params_upon_resume)(struct snd_sof_dev *sdev); /* optional */
-- 
cgit v1.2.3-59-g8ed1b


From 184fdfca4ba63d57a668564639a4dcf188fc9822 Mon Sep 17 00:00:00 2001
From: Keyon Jie <yang.jie@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:51 -0500
Subject: ASoC: SOF: Intel: hda: correct ROM state mask

The ROM state is represented by the 24 LSB bits in the ROM status
register, so the mask should be 0xffffff instead of 0xf.

Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 84d5fd021ccb..70909debfeb5 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -175,7 +175,7 @@
 #define HDA_DSP_STACK_DUMP_SIZE			32
 
 /* ROM  status/error values */
-#define HDA_DSP_ROM_STS_MASK			0xf
+#define HDA_DSP_ROM_STS_MASK			GENMASK(23, 0)
 #define HDA_DSP_ROM_INIT			0x1
 #define HDA_DSP_ROM_FW_MANIFEST_LOADED		0x3
 #define HDA_DSP_ROM_FW_FW_LOADED		0x4
-- 
cgit v1.2.3-59-g8ed1b


From 6aa232e1ccfcd2b55306b86803dc547c36f8ad15 Mon Sep 17 00:00:00 2001
From: Rander Wang <rander.wang@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:52 -0500
Subject: ASoC: SOF: Intel: hda: reduce ifdef usage for hda

Move the code for hda to one point

Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-dsp.c | 22 ++++++++++------------
 1 file changed, 10 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 4315896b3f07..3d711d354fb9 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -359,6 +359,16 @@ static int hda_resume(struct snd_sof_dev *sdev)
 		bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
 
 	hda_dsp_ctrl_misc_clock_gating(sdev, true);
+
+	/* turn off the links that were off before suspend */
+	list_for_each_entry(hlink, &bus->hlink_list, list) {
+		if (!hlink->ref_count)
+			snd_hdac_ext_bus_link_power_down(hlink);
+	}
+
+	/* check dma status and clean up CORB/RIRB buffers */
+	if (!bus->cmd_dma_state)
+		snd_hdac_bus_stop_cmd_io(bus);
 #else
 
 	hda_dsp_ctrl_misc_clock_gating(sdev, false);
@@ -391,18 +401,6 @@ static int hda_resume(struct snd_sof_dev *sdev)
 	hda_dsp_ctrl_ppcap_enable(sdev, true);
 	hda_dsp_ctrl_ppcap_int_enable(sdev, true);
 
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-	/* turn off the links that were off before suspend */
-	list_for_each_entry(hlink, &bus->hlink_list, list) {
-		if (!hlink->ref_count)
-			snd_hdac_ext_bus_link_power_down(hlink);
-	}
-
-	/* check dma status and clean up CORB/RIRB buffers */
-	if (!bus->cmd_dma_state)
-		snd_hdac_bus_stop_cmd_io(bus);
-#endif
-
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From fd15f2f5e272145269bcbf834e0e0b560a575891 Mon Sep 17 00:00:00 2001
From: Rander Wang <rander.wang@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:53 -0500
Subject: ASoC: SOF: Intel: hda: Enable jack detection

In commit 7d4f606c50ff ("ALSA: hda - WAKEEN feature enabling for
runtime pm"), legacy HD-A driver sets hda controller in reset mode after
entering runtime-suspend. And when resuming from suspend mode, it checks
hda controller & codec status to detect headphone hotplug event. Now
this patch does the same job in SOF runtime pm functions.

And we need to check all the non-hdmi codecs for some cases like playback
with HDMI or capture with DMIC connected to dsp. In these cases, only
controller is active and codecs are suspended, so codecs can't send
unsolicited event to controller. The jack polling operation will activate
codecs and unsolicited event can work even codecs become suspended later.

Tested on whiskylake with hda codecs.

Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-codec.c | 44 +++++++++++++++++++++++++++++++++++++++--
 sound/soc/sof/intel/hda-dsp.c   | 21 ++++++++++----------
 sound/soc/sof/intel/hda.h       |  2 ++
 3 files changed, 54 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 0d8437b080bf..3ca6795a89ba 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -10,6 +10,7 @@
 
 #include <linux/module.h>
 #include <sound/hdaudio_ext.h>
+#include <sound/hda_register.h>
 #include <sound/hda_codec.h>
 #include <sound/hda_i915.h>
 #include <sound/sof.h>
@@ -37,16 +38,55 @@ static void hda_codec_load_module(struct hda_codec *codec)
 static void hda_codec_load_module(struct hda_codec *codec) {}
 #endif
 
+/* enable controller wake up event for all codecs with jack connectors */
+void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev)
+{
+	struct hda_bus *hbus = sof_to_hbus(sdev);
+	struct hdac_bus *bus = sof_to_bus(sdev);
+	struct hda_codec *codec;
+	unsigned int mask = 0;
+
+	list_for_each_codec(codec, hbus)
+		if (codec->jacktbl.used)
+			mask |= BIT(codec->core.addr);
+
+	snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, mask);
+}
+
+/* check jack status after resuming from suspend mode */
+void hda_codec_jack_check(struct snd_sof_dev *sdev)
+{
+	struct hda_bus *hbus = sof_to_hbus(sdev);
+	struct hdac_bus *bus = sof_to_bus(sdev);
+	struct hda_codec *codec;
+
+	/* disable controller Wake Up event*/
+	snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, 0);
+
+	list_for_each_codec(codec, hbus)
+		/*
+		 * Wake up all jack-detecting codecs regardless whether an event
+		 * has been recorded in STATESTS
+		 */
+		if (codec->jacktbl.used)
+			schedule_delayed_work(&codec->jackpoll_work,
+					      codec->jackpoll_interval);
+}
+#else
+void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {}
+void hda_codec_jack_check(struct snd_sof_dev *sdev) {}
 #endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */
+EXPORT_SYMBOL(hda_codec_jack_wake_enable);
+EXPORT_SYMBOL(hda_codec_jack_check);
 
 /* probe individual codec */
 static int hda_codec_probe(struct snd_sof_dev *sdev, int address)
 {
-	struct hda_bus *hbus = sof_to_hbus(sdev);
-	struct hdac_device *hdev;
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
 	struct hdac_hda_priv *hda_priv;
 #endif
+	struct hda_bus *hbus = sof_to_hbus(sdev);
+	struct hdac_device *hdev;
 	u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) |
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	u32 resp = -1;
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 3d711d354fb9..f9579edbca68 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -295,6 +295,9 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
 	hda_dsp_ipc_int_disable(sdev);
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	if (runtime_suspend)
+		hda_codec_jack_wake_enable(sdev);
+
 	/* power down all hda link */
 	snd_hdac_ext_bus_link_power_down_all(bus);
 #endif
@@ -329,7 +332,7 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
 	return 0;
 }
 
-static int hda_resume(struct snd_sof_dev *sdev)
+static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume)
 {
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	struct hdac_bus *bus = sof_to_bus(sdev);
@@ -343,7 +346,6 @@ static int hda_resume(struct snd_sof_dev *sdev)
 	 */
 	snd_sof_pci_update_bits(sdev, PCI_TCSEL, 0x07, 0);
 
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	/* reset and start hda controller */
 	ret = hda_dsp_ctrl_init_chip(sdev, true);
 	if (ret < 0) {
@@ -352,13 +354,10 @@ static int hda_resume(struct snd_sof_dev *sdev)
 		return ret;
 	}
 
-	hda_dsp_ctrl_misc_clock_gating(sdev, false);
-
-	/* Reset stream-to-link mapping */
-	list_for_each_entry(hlink, &bus->hlink_list, list)
-		bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
-
-	hda_dsp_ctrl_misc_clock_gating(sdev, true);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	/* check jack status */
+	if (runtime_resume)
+		hda_codec_jack_check(sdev);
 
 	/* turn off the links that were off before suspend */
 	list_for_each_entry(hlink, &bus->hlink_list, list) {
@@ -407,13 +406,13 @@ static int hda_resume(struct snd_sof_dev *sdev)
 int hda_dsp_resume(struct snd_sof_dev *sdev)
 {
 	/* init hda controller. DSP cores will be powered up during fw boot */
-	return hda_resume(sdev);
+	return hda_resume(sdev, false);
 }
 
 int hda_dsp_runtime_resume(struct snd_sof_dev *sdev)
 {
 	/* init hda controller. DSP cores will be powered up during fw boot */
-	return hda_resume(sdev);
+	return hda_resume(sdev, true);
 }
 
 int hda_dsp_runtime_idle(struct snd_sof_dev *sdev)
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 70909debfeb5..028e865d5e20 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -557,6 +557,8 @@ void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev,
  * HDA Codec operations.
  */
 int hda_codec_probe_bus(struct snd_sof_dev *sdev);
+void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev);
+void hda_codec_jack_check(struct snd_sof_dev *sdev);
 
 #endif /* CONFIG_SND_SOC_SOF_HDA */
 
-- 
cgit v1.2.3-59-g8ed1b


From 061edb232505797aee53a955225af23fa30d45b8 Mon Sep 17 00:00:00 2001
From: Zhu Yingjiang <yingjiang.zhu@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:54 -0500
Subject: ASoC: SOF: Intel: hda: set position buffer in init chip

Set the HDA stream position buffer during init chip. The position buffer
needs to be set in both HDA codec and nocodec cases. Using SOF defined
function and move it to common code.

Signed-off-by: Zhu Yingjiang <yingjiang.zhu@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-ctrl.c | 9 ++++-----
 1 file changed, 4 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index ea63f83a509b..015760284969 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -245,14 +245,13 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
 				SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN,
 				SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN);
 
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	/* program the position buffer */
 	if (bus->use_posbuf && bus->posbuf.addr) {
-		snd_hdac_chip_writel(bus, DPLBASE, (u32)bus->posbuf.addr);
-		snd_hdac_chip_writel(bus, DPUBASE,
-				     upper_32_bits(bus->posbuf.addr));
+		snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_ADSP_DPLBASE,
+				  (u32)bus->posbuf.addr);
+		snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_ADSP_DPUBASE,
+				  upper_32_bits(bus->posbuf.addr));
 	}
-#endif
 
 	bus->chip_init = true;
 
-- 
cgit v1.2.3-59-g8ed1b


From d06973515f95ac74d68fb3e1f1fb4b055531296c Mon Sep 17 00:00:00 2001
From: Zhu Yingjiang <yingjiang.zhu@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:55 -0500
Subject: ASoC: SOF: Intel: hda: use SOF defined init chip in resume

Unify resume code by using SOF common function hda_dsp_ctrl_init_chip()
which can handle both HDA and non-HDA cases. Move code to reset
stream-to-link mapping into hda_dsp_ctrl_init_chip().

Signed-off-by: Zhu Yingjiang <yingjiang.zhu@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-15-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-ctrl.c |  9 +++++++++
 sound/soc/sof/intel/hda-dsp.c  | 26 --------------------------
 2 files changed, 9 insertions(+), 26 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 015760284969..8b856dc35e20 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -164,6 +164,9 @@ int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable)
 int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
 {
 	struct hdac_bus *bus = sof_to_bus(sdev);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	struct hdac_ext_link *hlink;
+#endif
 	struct hdac_stream *stream;
 	int sd_offset, ret = 0;
 
@@ -253,6 +256,12 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
 				  upper_32_bits(bus->posbuf.addr));
 	}
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	/* Reset stream-to-link mapping */
+	list_for_each_entry(hlink, &bus->hlink_list, list)
+		bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
+#endif
+
 	bus->chip_init = true;
 
 	hda_dsp_ctrl_misc_clock_gating(sdev, true);
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index f9579edbca68..f08a5d649346 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -368,32 +368,6 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume)
 	/* check dma status and clean up CORB/RIRB buffers */
 	if (!bus->cmd_dma_state)
 		snd_hdac_bus_stop_cmd_io(bus);
-#else
-
-	hda_dsp_ctrl_misc_clock_gating(sdev, false);
-
-	/* reset controller */
-	ret = hda_dsp_ctrl_link_reset(sdev, true);
-	if (ret < 0) {
-		dev_err(sdev->dev,
-			"error: failed to reset controller during resume\n");
-		return ret;
-	}
-
-	/* take controller out of reset */
-	ret = hda_dsp_ctrl_link_reset(sdev, false);
-	if (ret < 0) {
-		dev_err(sdev->dev,
-			"error: failed to ready controller during resume\n");
-		return ret;
-	}
-
-	/* enable hda bus irq */
-	snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL,
-				SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN,
-				SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN);
-
-	hda_dsp_ctrl_misc_clock_gating(sdev, true);
 #endif
 
 	/* enable ppcap interrupt */
-- 
cgit v1.2.3-59-g8ed1b


From dc7a36f178a94604d29c5dd15c77187905d8e882 Mon Sep 17 00:00:00 2001
From: Zhu Yingjiang <yingjiang.zhu@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:56 -0500
Subject: ASoC: SOF: Intel: hda: remove duplicated clear WAKESTS

Remove the first clear WAKESTS, only one clear is needed during init
chip.

Signed-off-by: Zhu Yingjiang <yingjiang.zhu@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-ctrl.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 8b856dc35e20..a7fee403cb90 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -176,11 +176,6 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
 	hda_dsp_ctrl_misc_clock_gating(sdev, false);
 
 	if (full_reset) {
-		/* clear WAKESTS */
-		snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS,
-					SOF_HDA_WAKESTS_INT_MASK,
-					SOF_HDA_WAKESTS_INT_MASK);
-
 		/* reset HDA controller */
 		ret = hda_dsp_ctrl_link_reset(sdev, true);
 		if (ret < 0) {
-- 
cgit v1.2.3-59-g8ed1b


From 672ff5e3596ee27b64edcc73251f4ae1c8ab12ac Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:57 -0500
Subject: ASoC: SOF: Intel: hda: add a parameter to disable MSI

Enabling MSI on HDA can fail, in which case the legacy PCI IRQ mode
will be used. To make testing this mode easier add an "enable_msi"
module parameter, which is only enabled if debugging is enabled too.

Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda.c | 28 +++++++++++++++++-----------
 sound/soc/sof/sof-priv.h  |  2 +-
 2 files changed, 18 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 7f665392618f..79cce20666b6 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -46,6 +46,12 @@ struct hda_dsp_msg_code {
 	const char *msg;
 };
 
+static bool hda_use_msi = IS_ENABLED(CONFIG_PCI);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG)
+module_param_named(use_msi, hda_use_msi, bool, 0444);
+MODULE_PARM_DESC(use_msi, "SOF HDA use PCI MSI mode");
+#endif
+
 static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = {
 	{HDA_DSP_ROM_FW_MANIFEST_LOADED, "status: manifest loaded"},
 	{HDA_DSP_ROM_FW_FW_LOADED, "status: fw loaded"},
@@ -529,11 +535,18 @@ int hda_dsp_probe(struct snd_sof_dev *sdev)
 	 * register our IRQ
 	 * let's try to enable msi firstly
 	 * if it fails, use legacy interrupt mode
-	 * TODO: support interrupt mode selection with kernel parameter
-	 *       support msi multiple vectors
+	 * TODO: support msi multiple vectors
 	 */
-	ret = pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI);
-	if (ret < 0) {
+	if (hda_use_msi && !pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI)) {
+		dev_info(sdev->dev, "use msi interrupt mode\n");
+		hdev->irq = pci_irq_vector(pci, 0);
+		/* ipc irq number is the same of hda irq */
+		sdev->ipc_irq = hdev->irq;
+		/* initialised to "false" by kzalloc() */
+		sdev->msi_enabled = true;
+	}
+
+	if (!sdev->msi_enabled) {
 		dev_info(sdev->dev, "use legacy interrupt mode\n");
 		/*
 		 * in IO-APIC mode, hda->irq and ipc_irq are using the same
@@ -541,13 +554,6 @@ int hda_dsp_probe(struct snd_sof_dev *sdev)
 		 */
 		hdev->irq = pci->irq;
 		sdev->ipc_irq = pci->irq;
-		sdev->msi_enabled = 0;
-	} else {
-		dev_info(sdev->dev, "use msi interrupt mode\n");
-		hdev->irq = pci_irq_vector(pci, 0);
-		/* ipc irq number is the same of hda irq */
-		sdev->ipc_irq = hdev->irq;
-		sdev->msi_enabled = 1;
 	}
 
 	dev_dbg(sdev->dev, "using HDA IRQ %d\n", hdev->irq);
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index b8c9274ccf42..983eadef4b30 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -435,7 +435,7 @@ struct snd_sof_dev {
 	u32 dtrace_error;
 	u32 dtrace_draining;
 
-	u32 msi_enabled;
+	bool msi_enabled;
 
 	void *private;			/* core does not touch this */
 };
-- 
cgit v1.2.3-59-g8ed1b


From a3ebccb52efdfb3459fa51516238cc464ee310c9 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:58 -0500
Subject: ASoC: SOF: Intel: hda: reset link DMA state in prepare

When application goes through SUSPEND/STOP->PREPARE->START
cycle, we should always reprogram the DAI link DMA to ensure
it is in sync with the host PCM DMA.

Use same state tracking logic to handle both restart and
system resume flows. Use link_prepared field of
'struct hdac_ext_stream' to store the state, instead of
adding redundant fields to SOF specific structs.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-dai.c |  8 +++-----
 sound/soc/sof/intel/hda-dsp.c | 17 ++++++-----------
 sound/soc/sof/intel/hda.h     |  1 -
 3 files changed, 9 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index a514f9cf5c9a..a448be60f6dd 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -226,8 +226,6 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream,
 
 	snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev);
 
-	hda_stream->hw_params_upon_resume = 0;
-
 	link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name);
 	if (!link)
 		return -EINVAL;
@@ -267,8 +265,7 @@ static int hda_link_pcm_prepare(struct snd_pcm_substream *substream,
 
 	hda_stream = hstream_to_sof_hda_stream(link_dev);
 
-	/* setup hw_params again only if resuming from system suspend */
-	if (!hda_stream->hw_params_upon_resume)
+	if (link_dev->link_prepared)
 		return 0;
 
 	dev_dbg(sdev->dev, "hda: prepare stream dir %d\n", substream->stream);
@@ -317,6 +314,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 		snd_hdac_ext_link_stream_start(link_dev);
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
 		/*
 		 * clear and release link DMA channel. It will be assigned when
 		 * hw_params is set up again after resume.
@@ -329,10 +327,10 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 		snd_hdac_ext_link_clear_stream_id(link, stream_tag);
 		snd_hdac_ext_stream_release(link_dev,
 					    HDAC_EXT_STREAM_TYPE_LINK);
+		link_dev->link_prepared = 0;
 
 		/* fallthrough */
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-	case SNDRV_PCM_TRIGGER_STOP:
 		snd_hdac_ext_link_stream_clear(link_dev);
 		break;
 	default:
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index f08a5d649346..e82ecaad1763 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -425,25 +425,19 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev)
 
 int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 {
-	struct hdac_bus *bus = sof_to_bus(sdev);
-	struct sof_intel_hda_stream *hda_stream;
-	struct hdac_ext_stream *stream;
-	struct hdac_stream *s;
-
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	struct hdac_bus *bus = sof_to_bus(sdev);
 	struct snd_soc_pcm_runtime *rtd;
+	struct hdac_ext_stream *stream;
 	struct hdac_ext_link *link;
+	struct hdac_stream *s;
 	const char *name;
 	int stream_tag;
-#endif
 
 	/* set internal flag for BE */
 	list_for_each_entry(s, &bus->stream_list, list) {
 		stream = stream_to_hdac_ext_stream(s);
-		hda_stream = container_of(stream, struct sof_intel_hda_stream,
-					  hda_stream);
-		hda_stream->hw_params_upon_resume = 1;
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+
 		/*
 		 * clear and release stream. This should already be taken care
 		 * for running streams when the SUSPEND trigger is called.
@@ -460,8 +454,9 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 			snd_hdac_ext_link_clear_stream_id(link, stream_tag);
 			snd_hdac_ext_stream_release(stream,
 						    HDAC_EXT_STREAM_TYPE_LINK);
+			stream->link_prepared = 0;
 		}
-#endif
 	}
+#endif
 	return 0;
 }
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 028e865d5e20..ceaaa8d467f4 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -418,7 +418,6 @@ struct sof_intel_hda_stream {
 	struct snd_sof_dev *sdev;
 	struct hdac_ext_stream hda_stream;
 	struct sof_intel_stream stream;
-	int hw_params_upon_resume; /* set up hw_params upon resume */
 	int host_reserved; /* reserve host DMA channel */
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 6298b78742be6593d372ed1b5bfa5397e1393595 Mon Sep 17 00:00:00 2001
From: Janusz Jankowski <janusz.jankowski@linux.intel.com>
Date: Mon, 22 Jul 2019 09:14:02 -0500
Subject: ASoC: SOF: Intel: ssp: BCLK delay parameter

Some codecs require BCLK to be on for some time, before sending
any data. SOF can enable BCLK and then wait for guaranteed time,
before starting DMA on SSP start.

Signed-off-by: Janusz Jankowski <janusz.jankowski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-22-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/sof/dai-intel.h   | 3 +++
 include/uapi/sound/sof/abi.h    | 2 +-
 include/uapi/sound/sof/tokens.h | 1 +
 sound/soc/sof/topology.c        | 3 +++
 4 files changed, 8 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h
index 4bb8ee138ba7..a81afd3fbd41 100644
--- a/include/sound/sof/dai-intel.h
+++ b/include/sound/sof/dai-intel.h
@@ -76,6 +76,9 @@ struct sof_ipc_dai_ssp_params {
 	uint16_t tdm_per_slot_padding_flag;
 	uint32_t clks_control;
 	uint32_t quirks;
+	uint32_t bclk_delay;	/* guaranteed time (ms) for which BCLK
+				 * will be driven, before sending data
+				 */
 } __packed;
 
 /* HDA Configuration Request - SOF_IPC_DAI_HDA_CONFIG */
diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h
index 4a9c24434f42..dff70a42445a 100644
--- a/include/uapi/sound/sof/abi.h
+++ b/include/uapi/sound/sof/abi.h
@@ -26,7 +26,7 @@
 
 /* SOF ABI version major, minor and patch numbers */
 #define SOF_ABI_MAJOR 3
-#define SOF_ABI_MINOR 8
+#define SOF_ABI_MINOR 9
 #define SOF_ABI_PATCH 0
 
 /* SOF ABI version number. Format within 32bit word is MMmmmppp */
diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h
index dc1b27daaac6..6435240cef13 100644
--- a/include/uapi/sound/sof/tokens.h
+++ b/include/uapi/sound/sof/tokens.h
@@ -75,6 +75,7 @@
 #define SOF_TKN_INTEL_SSP_FRAME_PULSE_WIDTH	503
 #define SOF_TKN_INTEL_SSP_QUIRKS		504
 #define SOF_TKN_INTEL_SSP_TDM_PADDING_PER_SLOT	505
+#define SOF_TKN_INTEL_SSP_BCLK_DELAY		506
 
 /* DMIC */
 #define SOF_TKN_INTEL_DMIC_DRIVER_VERSION	600
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 432ae343f960..12b7d900b9c2 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -748,6 +748,9 @@ static const struct sof_topology_token ssp_tokens[] = {
 		get_token_u16,
 		offsetof(struct sof_ipc_dai_ssp_params,
 			 tdm_per_slot_padding_flag), 0},
+	{SOF_TKN_INTEL_SSP_BCLK_DELAY, SND_SOC_TPLG_TUPLE_TYPE_WORD,
+		get_token_u32,
+		offsetof(struct sof_ipc_dai_ssp_params, bclk_delay), 0},
 
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 7ccafa2b3879612ab9ed88bb6664bd4085957186 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Thu, 11 Jul 2019 18:49:46 +0800
Subject: ASoC: fsl_esai: recover the channel swap after xrun

There is chip errata ERR008000, the reference doc is
(https://www.nxp.com/docs/en/errata/IMX6DQCE.pdf),

The issue is "While using ESAI transmit or receive and
an underrun/overrun happens, channel swap may occur.
The only recovery mechanism is to reset the ESAI."

This issue exist in imx3/imx5/imx6(partial) series.

In this commit add a tasklet to handle reset of ESAI
after xrun happens to recover the channel swap.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/326035cb99975361699d9ed748054b08bc06a341.1562842206.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_esai.c | 74 ++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 74 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index ab460d6d7432..4ce8ac769244 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -32,6 +32,7 @@
  * @extalclk: esai clock source to derive HCK, SCK and FS
  * @fsysclk: system clock source to derive HCK, SCK and FS
  * @spbaclk: SPBA clock (optional, depending on SoC design)
+ * @task: tasklet to handle the reset operation
  * @fifo_depth: depth of tx/rx FIFO
  * @slot_width: width of each DAI slot
  * @slots: number of slots
@@ -42,6 +43,7 @@
  * @sck_div: if using PSR/PM dividers for SCKx clock
  * @slave_mode: if fully using DAI slave mode
  * @synchronous: if using tx/rx synchronous mode
+ * @reset_at_xrun: flags for enable reset operaton
  * @name: driver name
  */
 struct fsl_esai {
@@ -53,6 +55,7 @@ struct fsl_esai {
 	struct clk *extalclk;
 	struct clk *fsysclk;
 	struct clk *spbaclk;
+	struct tasklet_struct task;
 	u32 fifo_depth;
 	u32 slot_width;
 	u32 slots;
@@ -65,6 +68,7 @@ struct fsl_esai {
 	bool sck_div[2];
 	bool slave_mode;
 	bool synchronous;
+	bool reset_at_xrun;
 	char name[32];
 };
 
@@ -73,8 +77,16 @@ static irqreturn_t esai_isr(int irq, void *devid)
 	struct fsl_esai *esai_priv = (struct fsl_esai *)devid;
 	struct platform_device *pdev = esai_priv->pdev;
 	u32 esr;
+	u32 saisr;
 
 	regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr);
+	regmap_read(esai_priv->regmap, REG_ESAI_SAISR, &saisr);
+
+	if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) &&
+	    esai_priv->reset_at_xrun) {
+		dev_dbg(&pdev->dev, "reset module for xrun\n");
+		tasklet_schedule(&esai_priv->task);
+	}
 
 	if (esr & ESAI_ESR_TINIT_MASK)
 		dev_dbg(&pdev->dev, "isr: Transmission Initialized\n");
@@ -635,10 +647,17 @@ static void fsl_esai_trigger_start(struct fsl_esai *esai_priv, bool tx)
 			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask));
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
 			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask));
+
+	/* Enable Exception interrupt */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+			   ESAI_xCR_xEIE_MASK, ESAI_xCR_xEIE);
 }
 
 static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx)
 {
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+			   ESAI_xCR_xEIE_MASK, 0);
+
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 			   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0);
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx),
@@ -653,6 +672,51 @@ static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx)
 			   ESAI_xFCR_xFR, 0);
 }
 
+static void fsl_esai_hw_reset(unsigned long arg)
+{
+	struct fsl_esai *esai_priv = (struct fsl_esai *)arg;
+	bool tx = true, rx = false, enabled[2];
+	u32 tfcr, rfcr;
+
+	/* Save the registers */
+	regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr);
+	regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr);
+	enabled[tx] = tfcr & ESAI_xFCR_xFEN;
+	enabled[rx] = rfcr & ESAI_xFCR_xFEN;
+
+	/* Stop the tx & rx */
+	fsl_esai_trigger_stop(esai_priv, tx);
+	fsl_esai_trigger_stop(esai_priv, rx);
+
+	/* Reset the esai, and ignore return value */
+	fsl_esai_hw_init(esai_priv);
+
+	/* Enforce ESAI personal resets for both TX and RX */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR,
+			   ESAI_xCR_xPR_MASK, ESAI_xCR_xPR);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR,
+			   ESAI_xCR_xPR_MASK, ESAI_xCR_xPR);
+
+	/* Restore registers by regcache_sync, and ignore return value */
+	fsl_esai_register_restore(esai_priv);
+
+	/* Remove ESAI personal resets by configuring PCRC and PRRC also */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR,
+			   ESAI_xCR_xPR_MASK, 0);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR,
+			   ESAI_xCR_xPR_MASK, 0);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
+			   ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
+			   ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
+
+	/* Restart tx / rx, if they already enabled */
+	if (enabled[tx])
+		fsl_esai_trigger_start(esai_priv, tx);
+	if (enabled[rx])
+		fsl_esai_trigger_start(esai_priv, rx);
+}
+
 static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 			    struct snd_soc_dai *dai)
 {
@@ -857,6 +921,10 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	esai_priv->pdev = pdev;
 	snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np);
 
+	if (of_device_is_compatible(np, "fsl,vf610-esai") ||
+	    of_device_is_compatible(np, "fsl,imx35-esai"))
+		esai_priv->reset_at_xrun = true;
+
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
@@ -956,6 +1024,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	tasklet_init(&esai_priv->task, fsl_esai_hw_reset,
+		     (unsigned long)esai_priv);
+
 	pm_runtime_enable(&pdev->dev);
 
 	regcache_cache_only(esai_priv->regmap, true);
@@ -969,7 +1040,10 @@ static int fsl_esai_probe(struct platform_device *pdev)
 
 static int fsl_esai_remove(struct platform_device *pdev)
 {
+	struct fsl_esai *esai_priv = platform_get_drvdata(pdev);
+
 	pm_runtime_disable(&pdev->dev);
+	tasklet_kill(&esai_priv->task);
 
 	return 0;
 }
-- 
cgit v1.2.3-59-g8ed1b


From 934bf82203d679a8f2ff22817e341bc059f5f7fc Mon Sep 17 00:00:00 2001
From: Rander Wang <rander.wang@linux.intel.com>
Date: Mon, 22 Jul 2019 09:13:59 -0500
Subject: ASoC: SOF: Intel: hda: fix link DMA config

For this bug, there are two capture pcm streams active, with one
stream and its related stream tag released before suspend. Later
when system suspend is done, the stream tag for the remaining
active stream is released by SOF driver. After system resume, hda
codec driver restores the stream tag for the active pcm stream,
but SOF goes to assign a new one, which now doesn't match with the
stream tag used by codec driver, and this causes DMA to fail
receiving data, leading to unrecoverable XRUN condition in FW.

For stream tag is stored in both hda codec and SOF driver, it
shouldn't be released only in SOF driver. This patch just keeps the
stream information in dma data and checks whether there is a stored
DMA data for stream resuming from S3 and restores it. And it also
removes DMA data when the stream is released.

Tested on Whiskey Lake platform.

GitHub issue: https://github.com/thesofproject/sof/issues/1594
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-dai.c | 15 +++++++++------
 sound/soc/sof/intel/hda-dsp.c | 10 ++++------
 2 files changed, 13 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index a448be60f6dd..2b5e2b8c69c2 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -210,9 +210,13 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream,
 	int stream_tag;
 	int ret;
 
-	link_dev = hda_link_stream_assign(bus, substream);
-	if (!link_dev)
-		return -EBUSY;
+	/* get stored dma data if resuming from system suspend */
+	link_dev = snd_soc_dai_get_dma_data(dai, substream);
+	if (!link_dev) {
+		link_dev = hda_link_stream_assign(bus, substream);
+		if (!link_dev)
+			return -EBUSY;
+	}
 
 	stream_tag = hdac_stream(link_dev)->stream_tag;
 
@@ -316,7 +320,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		/*
-		 * clear and release link DMA channel. It will be assigned when
+		 * clear link DMA channel. It will be assigned when
 		 * hw_params is set up again after resume.
 		 */
 		ret = hda_link_config_ipc(hda_stream, dai->name,
@@ -325,8 +329,6 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 			return ret;
 		stream_tag = hdac_stream(link_dev)->stream_tag;
 		snd_hdac_ext_link_clear_stream_id(link, stream_tag);
-		snd_hdac_ext_stream_release(link_dev,
-					    HDAC_EXT_STREAM_TYPE_LINK);
 		link_dev->link_prepared = 0;
 
 		/* fallthrough */
@@ -369,6 +371,7 @@ static int hda_link_hw_free(struct snd_pcm_substream *substream,
 
 	stream_tag = hdac_stream(link_dev)->stream_tag;
 	snd_hdac_ext_link_clear_stream_id(link, stream_tag);
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
 	snd_hdac_ext_stream_release(link_dev, HDAC_EXT_STREAM_TYPE_LINK);
 	link_dev->link_prepared = 0;
 
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index e82ecaad1763..e38008194574 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -439,10 +439,10 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 		stream = stream_to_hdac_ext_stream(s);
 
 		/*
-		 * clear and release stream. This should already be taken care
-		 * for running streams when the SUSPEND trigger is called.
-		 * But paused streams do not get suspended, so this needs to be
-		 * done explicitly during suspend.
+		 * clear stream. This should already be taken care for running
+		 * streams when the SUSPEND trigger is called. But paused
+		 * streams do not get suspended, so this needs to be done
+		 * explicitly during suspend.
 		 */
 		if (stream->link_substream) {
 			rtd = snd_pcm_substream_chip(stream->link_substream);
@@ -452,8 +452,6 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 				return -EINVAL;
 			stream_tag = hdac_stream(stream)->stream_tag;
 			snd_hdac_ext_link_clear_stream_id(link, stream_tag);
-			snd_hdac_ext_stream_release(stream,
-						    HDAC_EXT_STREAM_TYPE_LINK);
 			stream->link_prepared = 0;
 		}
 	}
-- 
cgit v1.2.3-59-g8ed1b


From 810dbea3656912d6ad8db691a9a4b2ad97d4b736 Mon Sep 17 00:00:00 2001
From: Rander Wang <rander.wang@linux.intel.com>
Date: Mon, 22 Jul 2019 09:14:00 -0500
Subject: ASoC: SOF: Intel: hda: fix stream id setting

snd_hdac_ext_link_clear_stream_id maps stream id to
link output, which is for playback, not capture.

Tested on Whiskey Lake platform.

Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-20-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-dai.c | 15 +++++++++++----
 sound/soc/sof/intel/hda-dsp.c |  8 +++++++-
 2 files changed, 18 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 2b5e2b8c69c2..8796f385be76 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -327,8 +327,12 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
 					  DMA_CHAN_INVALID, substream->stream);
 		if (ret < 0)
 			return ret;
-		stream_tag = hdac_stream(link_dev)->stream_tag;
-		snd_hdac_ext_link_clear_stream_id(link, stream_tag);
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			stream_tag = hdac_stream(link_dev)->stream_tag;
+			snd_hdac_ext_link_clear_stream_id(link, stream_tag);
+		}
+
 		link_dev->link_prepared = 0;
 
 		/* fallthrough */
@@ -369,8 +373,11 @@ static int hda_link_hw_free(struct snd_pcm_substream *substream,
 	if (!link)
 		return -EINVAL;
 
-	stream_tag = hdac_stream(link_dev)->stream_tag;
-	snd_hdac_ext_link_clear_stream_id(link, stream_tag);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		stream_tag = hdac_stream(link_dev)->stream_tag;
+		snd_hdac_ext_link_clear_stream_id(link, stream_tag);
+	}
+
 	snd_soc_dai_set_dma_data(dai, substream, NULL);
 	snd_hdac_ext_stream_release(link_dev, HDAC_EXT_STREAM_TYPE_LINK);
 	link_dev->link_prepared = 0;
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index e38008194574..fb55a3c5afd0 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -450,9 +450,15 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 			link = snd_hdac_ext_bus_get_link(bus, name);
 			if (!link)
 				return -EINVAL;
+
+			stream->link_prepared = 0;
+
+			if (hdac_stream(stream)->direction ==
+				SNDRV_PCM_STREAM_CAPTURE)
+				continue;
+
 			stream_tag = hdac_stream(stream)->stream_tag;
 			snd_hdac_ext_link_clear_stream_id(link, stream_tag);
-			stream->link_prepared = 0;
 		}
 	}
 #endif
-- 
cgit v1.2.3-59-g8ed1b


From f90aa354be7bffaec2b440eb1831c429ecb1a5e2 Mon Sep 17 00:00:00 2001
From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Date: Mon, 22 Jul 2019 23:26:39 +0200
Subject: ASoC: rt5665: Fix a typo in the name of a function

All function names start with rt5665_, except 'rt5655_set_verf()'.
It is likely a typo.

Fix it to be consistent.

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20190722212639.26954-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5665.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c
index c050d84a6916..68299ce26d3e 100644
--- a/sound/soc/codecs/rt5665.c
+++ b/sound/soc/codecs/rt5665.c
@@ -2566,7 +2566,7 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int rt5655_set_verf(struct snd_soc_dapm_widget *w,
+static int rt5665_set_verf(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
@@ -2686,11 +2686,11 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5665_PWR_VOL,
 		RT5665_PWR_MIC_DET_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Vref1", RT5665_PWR_ANLG_1, RT5665_PWR_VREF1_BIT, 0,
-		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("Vref2", RT5665_PWR_ANLG_1, RT5665_PWR_VREF2_BIT, 0,
-		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("Vref3", RT5665_PWR_ANLG_1, RT5665_PWR_VREF3_BIT, 0,
-		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 
 	/* ASRC */
 	SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5665_ASRC_1,
-- 
cgit v1.2.3-59-g8ed1b


From a789adae592e3d28ae87902c0012a8b11c96eb79 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Tue, 23 Jul 2019 18:12:47 +0800
Subject: ASoC: rt1308: Convert headers to SPDX

Convert include and source files to SPDX.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20190723101247.19741-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1308.c | 17 +++++++----------
 sound/soc/codecs/rt1308.h |  6 ++----
 2 files changed, 9 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
index cf3d72f99705..6f2ee6809dbb 100755
--- a/sound/soc/codecs/rt1308.c
+++ b/sound/soc/codecs/rt1308.c
@@ -1,13 +1,10 @@
-/*
- * rt1308.c  --  RT1308 ALSA SoC amplifier component driver
- *
- * Copyright 2019 Realtek Semiconductor Corp.
- * Author: Derek Fang <derek.fang@realtek.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// rt1308.c  --  RT1308 ALSA SoC amplifier component driver
+//
+// Copyright 2019 Realtek Semiconductor Corp.
+// Author: Derek Fang <derek.fang@realtek.com>
+//
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h
index c330aae1d527..ff7c423e879e 100755
--- a/sound/soc/codecs/rt1308.h
+++ b/sound/soc/codecs/rt1308.h
@@ -1,12 +1,10 @@
+/* SPDX-License-Identifier: GPL-2.0 */
 /*
- * RT1308.h  --  RT1308 ALSA SoC amplifier component driver
+ * rt1308.h  --  RT1308 ALSA SoC amplifier component driver
  *
  * Copyright 2019 Realtek Semiconductor Corp.
  * Author: Derek Fang <derek.fang@realtek.com>
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef _RT1308_H_
-- 
cgit v1.2.3-59-g8ed1b


From 859fd6cbf1fb32b5428c26f837215c085b8a822e Mon Sep 17 00:00:00 2001
From: Timo Wischer <twischer@de.adit-jv.com>
Date: Mon, 22 Jul 2019 16:24:01 +0900
Subject: ASoC: rsnd: Support hw_free() callback at DAI level

This patch provides the needed infrastructure to support calling hw_free()
at the DAI level. This is for example required to free resources allocated
in hw_params() callback.

The modification of __rsnd_mod_add_hw_params does not have any side
effects because rsnd_mod_ops::hw_params callback is not used by anyone
until now.

Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Link: https://lore.kernel.org/r/20190722072403.11008-2-jiada_wang@mentor.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/core.c | 16 +++++++++++++++-
 sound/soc/sh/rcar/rsnd.h | 12 +++++++++---
 2 files changed, 24 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 56e8dae9a15c..bda5b958d0dc 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1421,6 +1421,20 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream,
 					params_buffer_bytes(hw_params));
 }
 
+static int rsnd_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_dai *dai = rsnd_substream_to_dai(substream);
+	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
+	int ret;
+
+	ret = rsnd_dai_call(hw_free, io, substream);
+	if (ret)
+		return ret;
+
+	return snd_pcm_lib_free_pages(substream);
+}
+
 static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_dai *dai = rsnd_substream_to_dai(substream);
@@ -1436,7 +1450,7 @@ static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream)
 static const struct snd_pcm_ops rsnd_pcm_ops = {
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= rsnd_hw_params,
-	.hw_free	= snd_pcm_lib_free_pages,
+	.hw_free	= rsnd_hw_free,
 	.pointer	= rsnd_pointer,
 };
 
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 7727add3eb1a..ea6cbaa9743e 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -327,6 +327,9 @@ struct rsnd_mod_ops {
 	int (*cleanup)(struct rsnd_mod *mod,
 		       struct rsnd_dai_stream *io,
 		       struct rsnd_priv *priv);
+	int (*hw_free)(struct rsnd_mod *mod,
+		       struct rsnd_dai_stream *io,
+		       struct snd_pcm_substream *substream);
 	u32 *(*get_status)(struct rsnd_mod *mod,
 			   struct rsnd_dai_stream *io,
 			   enum rsnd_mod_type type);
@@ -351,12 +354,12 @@ struct rsnd_mod {
  *
  * B	0: init		1: quit
  * C	0: start	1: stop
+ * D	0: hw_params	1: hw_free
  *
  * H is always called (see __rsnd_mod_call)
  * H	0: probe	1: remove
  * H	0: pcm_new
  * H	0: fallback
- * H	0: hw_params
  * H	0: pointer
  * H	0: prepare
  * H	0: cleanup
@@ -365,12 +368,13 @@ struct rsnd_mod {
 #define __rsnd_mod_shift_quit		4
 #define __rsnd_mod_shift_start		8
 #define __rsnd_mod_shift_stop		8
+#define __rsnd_mod_shift_hw_params	12
+#define __rsnd_mod_shift_hw_free	12
 #define __rsnd_mod_shift_probe		28 /* always called */
 #define __rsnd_mod_shift_remove		28 /* always called */
 #define __rsnd_mod_shift_irq		28 /* always called */
 #define __rsnd_mod_shift_pcm_new	28 /* always called */
 #define __rsnd_mod_shift_fallback	28 /* always called */
-#define __rsnd_mod_shift_hw_params	28 /* always called */
 #define __rsnd_mod_shift_pointer	28 /* always called */
 #define __rsnd_mod_shift_prepare	28 /* always called */
 #define __rsnd_mod_shift_cleanup	28 /* always called */
@@ -383,10 +387,11 @@ struct rsnd_mod {
 #define __rsnd_mod_add_quit		-1
 #define __rsnd_mod_add_start		 1
 #define __rsnd_mod_add_stop		-1
+#define __rsnd_mod_add_hw_params	1
+#define __rsnd_mod_add_hw_free		-1
 #define __rsnd_mod_add_irq		0
 #define __rsnd_mod_add_pcm_new		0
 #define __rsnd_mod_add_fallback		0
-#define __rsnd_mod_add_hw_params	0
 #define __rsnd_mod_add_pointer		0
 
 #define __rsnd_mod_call_probe		0
@@ -402,6 +407,7 @@ struct rsnd_mod {
 #define __rsnd_mod_call_fallback	0
 #define __rsnd_mod_call_hw_params	0
 #define __rsnd_mod_call_pointer		0
+#define __rsnd_mod_call_hw_free		1
 
 #define rsnd_mod_to_priv(mod)	((mod)->priv)
 #define rsnd_mod_power_on(mod)	clk_enable((mod)->clk)
-- 
cgit v1.2.3-59-g8ed1b


From 06f6e1d41427f394ad3f67ecf06efcd28a46932c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:32:12 +0900
Subject: ASoC: add soc-dai.c

Current ALSA SoC has many snd_soc_dai_xxx() function which is
using dai->driver->ops->xxx.
But, some of them are implemented as snd_soc_dai_xxx(),
but others are directly using dai->driver->ops->xxx.
Because of it, the code is not easy to read.

This patch creats new soc-dai.c and moves snd_soc_dai_xxx()
functions into it.
One exception is snd_soc_dai_is_dummy() which is based on
soc-utils local variable. We need to keep it as-is there.

Others which is directly using dai->driver->ops->xxx will be
implemented at soc-dai.c by incremental patches.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871ryij1r6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/Makefile   |   2 +-
 sound/soc/soc-core.c | 243 ------------------------------------------------
 sound/soc/soc-dai.c  | 254 +++++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 255 insertions(+), 244 deletions(-)
 create mode 100644 sound/soc/soc-dai.c

(limited to 'sound')

diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index d90ce8a32887..919c3c027c62 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,5 @@
 # SPDX-License-Identifier: GPL-2.0
-snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o
 snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o
 snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fd6eaae6c0ed..6e8c5c8eeaec 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2394,26 +2394,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
 }
 EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls);
 
-/**
- * snd_soc_dai_set_sysclk - configure DAI system or master clock.
- * @dai: DAI
- * @clk_id: DAI specific clock ID
- * @freq: new clock frequency in Hz
- * @dir: new clock direction - input/output.
- *
- * Configures the DAI master (MCLK) or system (SYSCLK) clocking.
- */
-int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
-	unsigned int freq, int dir)
-{
-	if (dai->driver->ops->set_sysclk)
-		return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
-
-	return snd_soc_component_set_sysclk(dai->component, clk_id, 0,
-					    freq, dir);
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
-
 /**
  * snd_soc_component_set_sysclk - configure COMPONENT system or master clock.
  * @component: COMPONENT
@@ -2436,48 +2416,6 @@ int snd_soc_component_set_sysclk(struct snd_soc_component *component,
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_set_sysclk);
 
-/**
- * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
- * @dai: DAI
- * @div_id: DAI specific clock divider ID
- * @div: new clock divisor.
- *
- * Configures the clock dividers. This is used to derive the best DAI bit and
- * frame clocks from the system or master clock. It's best to set the DAI bit
- * and frame clocks as low as possible to save system power.
- */
-int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
-	int div_id, int div)
-{
-	if (dai->driver->ops->set_clkdiv)
-		return dai->driver->ops->set_clkdiv(dai, div_id, div);
-	else
-		return -EINVAL;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
-
-/**
- * snd_soc_dai_set_pll - configure DAI PLL.
- * @dai: DAI
- * @pll_id: DAI specific PLL ID
- * @source: DAI specific source for the PLL
- * @freq_in: PLL input clock frequency in Hz
- * @freq_out: requested PLL output clock frequency in Hz
- *
- * Configures and enables PLL to generate output clock based on input clock.
- */
-int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
-	unsigned int freq_in, unsigned int freq_out)
-{
-	if (dai->driver->ops->set_pll)
-		return dai->driver->ops->set_pll(dai, pll_id, source,
-					 freq_in, freq_out);
-
-	return snd_soc_component_set_pll(dai->component, pll_id, source,
-					 freq_in, freq_out);
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
-
 /*
  * snd_soc_component_set_pll - configure component PLL.
  * @component: COMPONENT
@@ -2500,187 +2438,6 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_set_pll);
 
-/**
- * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio.
- * @dai: DAI
- * @ratio: Ratio of BCLK to Sample rate.
- *
- * Configures the DAI for a preset BCLK to sample rate ratio.
- */
-int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
-{
-	if (dai->driver->ops->set_bclk_ratio)
-		return dai->driver->ops->set_bclk_ratio(dai, ratio);
-	else
-		return -EINVAL;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
-
-/**
- * snd_soc_dai_set_fmt - configure DAI hardware audio format.
- * @dai: DAI
- * @fmt: SND_SOC_DAIFMT_* format value.
- *
- * Configures the DAI hardware format and clocking.
- */
-int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
-{
-	if (dai->driver->ops->set_fmt == NULL)
-		return -ENOTSUPP;
-	return dai->driver->ops->set_fmt(dai, fmt);
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
-
-/**
- * snd_soc_xlate_tdm_slot - generate tx/rx slot mask.
- * @slots: Number of slots in use.
- * @tx_mask: bitmask representing active TX slots.
- * @rx_mask: bitmask representing active RX slots.
- *
- * Generates the TDM tx and rx slot default masks for DAI.
- */
-static int snd_soc_xlate_tdm_slot_mask(unsigned int slots,
-				       unsigned int *tx_mask,
-				       unsigned int *rx_mask)
-{
-	if (*tx_mask || *rx_mask)
-		return 0;
-
-	if (!slots)
-		return -EINVAL;
-
-	*tx_mask = (1 << slots) - 1;
-	*rx_mask = (1 << slots) - 1;
-
-	return 0;
-}
-
-/**
- * snd_soc_dai_set_tdm_slot() - Configures a DAI for TDM operation
- * @dai: The DAI to configure
- * @tx_mask: bitmask representing active TX slots.
- * @rx_mask: bitmask representing active RX slots.
- * @slots: Number of slots in use.
- * @slot_width: Width in bits for each slot.
- *
- * This function configures the specified DAI for TDM operation. @slot contains
- * the total number of slots of the TDM stream and @slot_with the width of each
- * slot in bit clock cycles. @tx_mask and @rx_mask are bitmasks specifying the
- * active slots of the TDM stream for the specified DAI, i.e. which slots the
- * DAI should write to or read from. If a bit is set the corresponding slot is
- * active, if a bit is cleared the corresponding slot is inactive. Bit 0 maps to
- * the first slot, bit 1 to the second slot and so on. The first active slot
- * maps to the first channel of the DAI, the second active slot to the second
- * channel and so on.
- *
- * TDM mode can be disabled by passing 0 for @slots. In this case @tx_mask,
- * @rx_mask and @slot_width will be ignored.
- *
- * Returns 0 on success, a negative error code otherwise.
- */
-int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
-	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
-{
-	if (dai->driver->ops->xlate_tdm_slot_mask)
-		dai->driver->ops->xlate_tdm_slot_mask(slots,
-						&tx_mask, &rx_mask);
-	else
-		snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
-
-	dai->tx_mask = tx_mask;
-	dai->rx_mask = rx_mask;
-
-	if (dai->driver->ops->set_tdm_slot)
-		return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask,
-				slots, slot_width);
-	else
-		return -ENOTSUPP;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
-
-/**
- * snd_soc_dai_set_channel_map - configure DAI audio channel map
- * @dai: DAI
- * @tx_num: how many TX channels
- * @tx_slot: pointer to an array which imply the TX slot number channel
- *           0~num-1 uses
- * @rx_num: how many RX channels
- * @rx_slot: pointer to an array which imply the RX slot number channel
- *           0~num-1 uses
- *
- * configure the relationship between channel number and TDM slot number.
- */
-int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
-	unsigned int tx_num, unsigned int *tx_slot,
-	unsigned int rx_num, unsigned int *rx_slot)
-{
-	if (dai->driver->ops->set_channel_map)
-		return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot,
-			rx_num, rx_slot);
-	else
-		return -ENOTSUPP;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
-
-/**
- * snd_soc_dai_get_channel_map - Get DAI audio channel map
- * @dai: DAI
- * @tx_num: how many TX channels
- * @tx_slot: pointer to an array which imply the TX slot number channel
- *           0~num-1 uses
- * @rx_num: how many RX channels
- * @rx_slot: pointer to an array which imply the RX slot number channel
- *           0~num-1 uses
- */
-int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
-	unsigned int *tx_num, unsigned int *tx_slot,
-	unsigned int *rx_num, unsigned int *rx_slot)
-{
-	if (dai->driver->ops->get_channel_map)
-		return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot,
-			rx_num, rx_slot);
-	else
-		return -ENOTSUPP;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map);
-
-/**
- * snd_soc_dai_set_tristate - configure DAI system or master clock.
- * @dai: DAI
- * @tristate: tristate enable
- *
- * Tristates the DAI so that others can use it.
- */
-int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
-{
-	if (dai->driver->ops->set_tristate)
-		return dai->driver->ops->set_tristate(dai, tristate);
-	else
-		return -EINVAL;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
-
-/**
- * snd_soc_dai_digital_mute - configure DAI system or master clock.
- * @dai: DAI
- * @mute: mute enable
- * @direction: stream to mute
- *
- * Mutes the DAI DAC.
- */
-int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
-			     int direction)
-{
-	if (dai->driver->ops->mute_stream)
-		return dai->driver->ops->mute_stream(dai, mute, direction);
-	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
-		 dai->driver->ops->digital_mute)
-		return dai->driver->ops->digital_mute(dai, mute);
-	else
-		return -ENOTSUPP;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
-
 static int snd_soc_bind_card(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
new file mode 100644
index 000000000000..a1009ead40de
--- /dev/null
+++ b/sound/soc/soc-dai.c
@@ -0,0 +1,254 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// soc-dai.c
+//
+// Copyright (C) 2019 Renesas Electronics Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+/**
+ * snd_soc_dai_set_sysclk - configure DAI system or master clock.
+ * @dai: DAI
+ * @clk_id: DAI specific clock ID
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the DAI master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			   unsigned int freq, int dir)
+{
+	if (dai->driver->ops->set_sysclk)
+		return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
+
+	return snd_soc_component_set_sysclk(dai->component, clk_id, 0,
+					    freq, dir);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
+
+/**
+ * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
+ * @dai: DAI
+ * @div_id: DAI specific clock divider ID
+ * @div: new clock divisor.
+ *
+ * Configures the clock dividers. This is used to derive the best DAI bit and
+ * frame clocks from the system or master clock. It's best to set the DAI bit
+ * and frame clocks as low as possible to save system power.
+ */
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+			   int div_id, int div)
+{
+	if (dai->driver->ops->set_clkdiv)
+		return dai->driver->ops->set_clkdiv(dai, div_id, div);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
+
+/**
+ * snd_soc_dai_set_pll - configure DAI PLL.
+ * @dai: DAI
+ * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+			unsigned int freq_in, unsigned int freq_out)
+{
+	if (dai->driver->ops->set_pll)
+		return dai->driver->ops->set_pll(dai, pll_id, source,
+						 freq_in, freq_out);
+
+	return snd_soc_component_set_pll(dai->component, pll_id, source,
+					 freq_in, freq_out);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
+
+/**
+ * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio.
+ * @dai: DAI
+ * @ratio: Ratio of BCLK to Sample rate.
+ *
+ * Configures the DAI for a preset BCLK to sample rate ratio.
+ */
+int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	if (dai->driver->ops->set_bclk_ratio)
+		return dai->driver->ops->set_bclk_ratio(dai, ratio);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
+
+/**
+ * snd_soc_dai_set_fmt - configure DAI hardware audio format.
+ * @dai: DAI
+ * @fmt: SND_SOC_DAIFMT_* format value.
+ *
+ * Configures the DAI hardware format and clocking.
+ */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	if (dai->driver->ops->set_fmt == NULL)
+		return -ENOTSUPP;
+	return dai->driver->ops->set_fmt(dai, fmt);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
+
+/**
+ * snd_soc_xlate_tdm_slot - generate tx/rx slot mask.
+ * @slots: Number of slots in use.
+ * @tx_mask: bitmask representing active TX slots.
+ * @rx_mask: bitmask representing active RX slots.
+ *
+ * Generates the TDM tx and rx slot default masks for DAI.
+ */
+static int snd_soc_xlate_tdm_slot_mask(unsigned int slots,
+				       unsigned int *tx_mask,
+				       unsigned int *rx_mask)
+{
+	if (*tx_mask || *rx_mask)
+		return 0;
+
+	if (!slots)
+		return -EINVAL;
+
+	*tx_mask = (1 << slots) - 1;
+	*rx_mask = (1 << slots) - 1;
+
+	return 0;
+}
+
+/**
+ * snd_soc_dai_set_tdm_slot() - Configures a DAI for TDM operation
+ * @dai: The DAI to configure
+ * @tx_mask: bitmask representing active TX slots.
+ * @rx_mask: bitmask representing active RX slots.
+ * @slots: Number of slots in use.
+ * @slot_width: Width in bits for each slot.
+ *
+ * This function configures the specified DAI for TDM operation. @slot contains
+ * the total number of slots of the TDM stream and @slot_with the width of each
+ * slot in bit clock cycles. @tx_mask and @rx_mask are bitmasks specifying the
+ * active slots of the TDM stream for the specified DAI, i.e. which slots the
+ * DAI should write to or read from. If a bit is set the corresponding slot is
+ * active, if a bit is cleared the corresponding slot is inactive. Bit 0 maps to
+ * the first slot, bit 1 to the second slot and so on. The first active slot
+ * maps to the first channel of the DAI, the second active slot to the second
+ * channel and so on.
+ *
+ * TDM mode can be disabled by passing 0 for @slots. In this case @tx_mask,
+ * @rx_mask and @slot_width will be ignored.
+ *
+ * Returns 0 on success, a negative error code otherwise.
+ */
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+			     unsigned int tx_mask, unsigned int rx_mask,
+			     int slots, int slot_width)
+{
+	if (dai->driver->ops->xlate_tdm_slot_mask)
+		dai->driver->ops->xlate_tdm_slot_mask(slots,
+						      &tx_mask, &rx_mask);
+	else
+		snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
+
+	dai->tx_mask = tx_mask;
+	dai->rx_mask = rx_mask;
+
+	if (dai->driver->ops->set_tdm_slot)
+		return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask,
+						      slots, slot_width);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
+
+/**
+ * snd_soc_dai_set_channel_map - configure DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ *           0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ *           0~num-1 uses
+ *
+ * configure the relationship between channel number and TDM slot number.
+ */
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+				unsigned int tx_num, unsigned int *tx_slot,
+				unsigned int rx_num, unsigned int *rx_slot)
+{
+	if (dai->driver->ops->set_channel_map)
+		return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot,
+							 rx_num, rx_slot);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
+
+/**
+ * snd_soc_dai_get_channel_map - Get DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ *           0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ *           0~num-1 uses
+ */
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+				unsigned int *tx_num, unsigned int *tx_slot,
+				unsigned int *rx_num, unsigned int *rx_slot)
+{
+	if (dai->driver->ops->get_channel_map)
+		return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot,
+							 rx_num, rx_slot);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map);
+
+/**
+ * snd_soc_dai_set_tristate - configure DAI system or master clock.
+ * @dai: DAI
+ * @tristate: tristate enable
+ *
+ * Tristates the DAI so that others can use it.
+ */
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+	if (dai->driver->ops->set_tristate)
+		return dai->driver->ops->set_tristate(dai, tristate);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
+
+/**
+ * snd_soc_dai_digital_mute - configure DAI system or master clock.
+ * @dai: DAI
+ * @mute: mute enable
+ * @direction: stream to mute
+ *
+ * Mutes the DAI DAC.
+ */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+			     int direction)
+{
+	if (dai->driver->ops->mute_stream)
+		return dai->driver->ops->mute_stream(dai, mute, direction);
+	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
+		 dai->driver->ops->digital_mute)
+		return dai->driver->ops->digital_mute(dai, mute);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
-- 
cgit v1.2.3-59-g8ed1b


From aa6166c2ac28392d64f2d8b3acfb56c8fe657147 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:33:04 +0900
Subject: ASoC: soc-dai: mv soc_dai_hw_params() to soc-dai

Sometimes ALSA SoC naming is very random.
Current soc_dai_hw_params() should use snd_soc_dai_xxx() style.
And then, 1st parameter should be dai. Otherwise it is confusable.
 - soc_dai_hw_params(..., dai);
 + snd_soc_dai_hw_params(dai, ...);

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhl6hn5b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  4 ++++
 include/sound/soc.h     |  4 ----
 sound/soc/soc-dai.c     | 30 ++++++++++++++++++++++++++++++
 sound/soc/soc-dapm.c    |  4 ++--
 sound/soc/soc-pcm.c     | 35 +++--------------------------------
 5 files changed, 39 insertions(+), 38 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index f5d70041108f..3773262a1b77 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -145,6 +145,10 @@ int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
 
 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 
+int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
+			  struct snd_pcm_substream *substream,
+			  struct snd_pcm_hw_params *params);
+
 struct snd_soc_dai_ops {
 	/*
 	 * DAI clocking configuration, all optional.
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4e8071269639..d770606732cd 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -505,10 +505,6 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
 	const struct snd_pcm_hardware *hw);
 
-int soc_dai_hw_params(struct snd_pcm_substream *substream,
-		      struct snd_pcm_hw_params *params,
-		      struct snd_soc_dai *dai);
-
 /* Jack reporting */
 int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type,
 	struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins,
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index a1009ead40de..f883d27d136f 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -252,3 +252,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 		return -ENOTSUPP;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
+
+int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
+			  struct snd_pcm_substream *substream,
+			  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int ret;
+
+	/* perform any topology hw_params fixups before DAI  */
+	if (rtd->dai_link->be_hw_params_fixup) {
+		ret = rtd->dai_link->be_hw_params_fixup(rtd, params);
+		if (ret < 0) {
+			dev_err(rtd->dev,
+				"ASoC: hw_params topology fixup failed %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	if (dai->driver->ops->hw_params) {
+		ret = dai->driver->ops->hw_params(substream, params, dai);
+		if (ret < 0) {
+			dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n",
+				dai->name, ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f013b24c050a..8fc6a01f5d8b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3839,7 +3839,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				}
 			}
 			source->active++;
-			ret = soc_dai_hw_params(&substream, params, source);
+			ret = snd_soc_dai_hw_params(source, &substream, params);
 			if (ret < 0)
 				goto out;
 
@@ -3861,7 +3861,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				}
 			}
 			sink->active++;
-			ret = soc_dai_hw_params(&substream, params, sink);
+			ret = snd_soc_dai_hw_params(sink, &substream, params);
 			if (ret < 0)
 				goto out;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 4878d22ebd8c..420cc94e0a46 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -877,36 +877,6 @@ static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params,
 	interval->max = channels;
 }
 
-int soc_dai_hw_params(struct snd_pcm_substream *substream,
-		      struct snd_pcm_hw_params *params,
-		      struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int ret;
-
-	/* perform any topology hw_params fixups before DAI  */
-	if (rtd->dai_link->be_hw_params_fixup) {
-		ret = rtd->dai_link->be_hw_params_fixup(rtd, params);
-		if (ret < 0) {
-			dev_err(rtd->dev,
-				"ASoC: hw_params topology fixup failed %d\n",
-				ret);
-			return ret;
-		}
-	}
-
-	if (dai->driver->ops->hw_params) {
-		ret = dai->driver->ops->hw_params(substream, params, dai);
-		if (ret < 0) {
-			dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n",
-				dai->name, ret);
-			return ret;
-		}
-	}
-
-	return 0;
-}
-
 static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream,
 				      struct snd_soc_component *last)
 {
@@ -989,7 +959,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 			soc_pcm_codec_params_fixup(&codec_params,
 						   codec_dai->rx_mask);
 
-		ret = soc_dai_hw_params(substream, &codec_params, codec_dai);
+		ret = snd_soc_dai_hw_params(codec_dai, substream,
+					    &codec_params);
 		if(ret < 0)
 			goto codec_err;
 
@@ -1001,7 +972,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		snd_soc_dapm_update_dai(substream, &codec_params, codec_dai);
 	}
 
-	ret = soc_dai_hw_params(substream, params, cpu_dai);
+	ret = snd_soc_dai_hw_params(cpu_dai, substream, params);
 	if (ret < 0)
 		goto interface_err;
 
-- 
cgit v1.2.3-59-g8ed1b


From 846faaed9df7899e74311db3aec0a41a2f6bc345 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:33:19 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_hw_free()

Current ALSA SoC is directly using dai->driver->ops->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_hw_free() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y30qhn4w.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-dai.c     |  7 +++++++
 sound/soc/soc-dapm.c    |  7 ++-----
 sound/soc/soc-pcm.c     | 12 ++++--------
 4 files changed, 15 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 3773262a1b77..5222b6a758f2 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -148,6 +148,8 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
 			  struct snd_pcm_substream *substream,
 			  struct snd_pcm_hw_params *params);
+void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
+			 struct snd_pcm_substream *substream);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index f883d27d136f..39a685e6acd5 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -282,3 +282,10 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
 
 	return 0;
 }
+
+void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
+			 struct snd_pcm_substream *substream)
+{
+	if (dai->driver->ops->hw_free)
+		dai->driver->ops->hw_free(substream, dai);
+}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8fc6a01f5d8b..0783b05133ad 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3898,9 +3898,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
 
-			if (source->driver->ops->hw_free)
-				source->driver->ops->hw_free(&substream,
-							     source);
+			snd_soc_dai_hw_free(source, &substream);
 
 			source->active--;
 			if (source->driver->ops->shutdown)
@@ -3912,8 +3910,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
 
-			if (sink->driver->ops->hw_free)
-				sink->driver->ops->hw_free(&substream, sink);
+			snd_soc_dai_hw_free(sink, &substream);
 
 			sink->active--;
 			if (sink->driver->ops->shutdown)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 420cc94e0a46..58fc4e98ab59 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1011,8 +1011,7 @@ out:
 component_err:
 	soc_pcm_components_hw_free(substream, component);
 
-	if (cpu_dai->driver->ops->hw_free)
-		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+	snd_soc_dai_hw_free(cpu_dai, substream);
 	cpu_dai->rate = 0;
 
 interface_err:
@@ -1023,8 +1022,7 @@ codec_err:
 		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
 			continue;
 
-		if (codec_dai->driver->ops->hw_free)
-			codec_dai->driver->ops->hw_free(substream, codec_dai);
+		snd_soc_dai_hw_free(codec_dai, substream);
 		codec_dai->rate = 0;
 	}
 
@@ -1083,12 +1081,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
 			continue;
 
-		if (codec_dai->driver->ops->hw_free)
-			codec_dai->driver->ops->hw_free(substream, codec_dai);
+		snd_soc_dai_hw_free(codec_dai, substream);
 	}
 
-	if (cpu_dai->driver->ops->hw_free)
-		cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+	snd_soc_dai_hw_free(cpu_dai, substream);
 
 	mutex_unlock(&rtd->pcm_mutex);
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From 5a52a04531486e2ab069b7882432c8b266db36e6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:33:32 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_startup()

Current ALSA SoC is directly using dai->driver->ops->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_startup() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wogahn4i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-dai.c     | 11 +++++++++++
 sound/soc/soc-dapm.c    | 28 ++++++++++------------------
 sound/soc/soc-pcm.c     | 27 +++++++++++----------------
 4 files changed, 34 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 5222b6a758f2..0d16c5bb20bb 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -150,6 +150,8 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
 			  struct snd_pcm_hw_params *params);
 void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
 			 struct snd_pcm_substream *substream);
+int snd_soc_dai_startup(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 39a685e6acd5..6e196636e42f 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -289,3 +289,14 @@ void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
 	if (dai->driver->ops->hw_free)
 		dai->driver->ops->hw_free(substream, dai);
 }
+
+int snd_soc_dai_startup(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->startup)
+		ret = dai->driver->ops->startup(substream, dai);
+
+	return ret;
+}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0783b05133ad..71bfd049480a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3828,15 +3828,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
 
-			if (source->driver->ops->startup) {
-				ret = source->driver->ops->startup(&substream,
-								   source);
-				if (ret < 0) {
-					dev_err(source->dev,
-						"ASoC: startup() failed: %d\n",
-						ret);
-					goto out;
-				}
+			ret = snd_soc_dai_startup(source, &substream);
+			if (ret < 0) {
+				dev_err(source->dev,
+					"ASoC: startup() failed: %d\n", ret);
+				goto out;
 			}
 			source->active++;
 			ret = snd_soc_dai_hw_params(source, &substream, params);
@@ -3850,15 +3846,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
 
-			if (sink->driver->ops->startup) {
-				ret = sink->driver->ops->startup(&substream,
-								 sink);
-				if (ret < 0) {
-					dev_err(sink->dev,
-						"ASoC: startup() failed: %d\n",
-						ret);
-					goto out;
-				}
+			ret = snd_soc_dai_startup(sink, &substream);
+			if (ret < 0) {
+				dev_err(sink->dev,
+					"ASoC: startup() failed: %d\n", ret);
+				goto out;
 			}
 			sink->active++;
 			ret = snd_soc_dai_hw_params(sink, &substream, params);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 58fc4e98ab59..9c8713a3eef1 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -535,13 +535,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
 	/* startup the audio subsystem */
-	if (cpu_dai->driver->ops->startup) {
-		ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
-		if (ret < 0) {
-			dev_err(cpu_dai->dev, "ASoC: can't open interface"
-				" %s: %d\n", cpu_dai->name, ret);
-			goto out;
-		}
+	ret = snd_soc_dai_startup(cpu_dai, substream);
+	if (ret < 0) {
+		dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n",
+			cpu_dai->name, ret);
+		goto out;
 	}
 
 	ret = soc_pcm_components_open(substream, &component);
@@ -549,15 +547,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		goto component_err;
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->driver->ops->startup) {
-			ret = codec_dai->driver->ops->startup(substream,
-							      codec_dai);
-			if (ret < 0) {
-				dev_err(codec_dai->dev,
-					"ASoC: can't open codec %s: %d\n",
-					codec_dai->name, ret);
-				goto codec_dai_err;
-			}
+		ret = snd_soc_dai_startup(codec_dai, substream);
+		if (ret < 0) {
+			dev_err(codec_dai->dev,
+				"ASoC: can't open codec %s: %d\n",
+				codec_dai->name, ret);
+			goto codec_dai_err;
 		}
 
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-- 
cgit v1.2.3-59-g8ed1b


From 330fcb5135e0588b1ea3b0bbab587d1317c1cf7b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:33:39 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_shutdown()

Current ALSA SoC is directly using dai->driver->ops->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_shutdown() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v9vuhn4b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-dai.c     |  7 +++++++
 sound/soc/soc-dapm.c    |  7 ++-----
 sound/soc/soc-pcm.c     | 18 ++++++------------
 4 files changed, 17 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 0d16c5bb20bb..32545d457b3d 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -152,6 +152,8 @@ void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
 			 struct snd_pcm_substream *substream);
 int snd_soc_dai_startup(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream);
+void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
+			  struct snd_pcm_substream *substream);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 6e196636e42f..67ff6cc1fe02 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -300,3 +300,10 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
 
 	return ret;
 }
+
+void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
+			 struct snd_pcm_substream *substream)
+{
+	if (dai->driver->ops->shutdown)
+		dai->driver->ops->shutdown(substream, dai);
+}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 71bfd049480a..1d04612601ad 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3893,9 +3893,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 			snd_soc_dai_hw_free(source, &substream);
 
 			source->active--;
-			if (source->driver->ops->shutdown)
-				source->driver->ops->shutdown(&substream,
-							      source);
+			snd_soc_dai_shutdown(source, &substream);
 		}
 
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
@@ -3905,8 +3903,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 			snd_soc_dai_hw_free(sink, &substream);
 
 			sink->active--;
-			if (sink->driver->ops->shutdown)
-				sink->driver->ops->shutdown(&substream, sink);
+			snd_soc_dai_shutdown(sink, &substream);
 		}
 		break;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 9c8713a3eef1..ed5ae23c7104 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -641,16 +641,13 @@ machine_err:
 	i = rtd->num_codecs;
 
 codec_dai_err:
-	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
-		if (codec_dai->driver->ops->shutdown)
-			codec_dai->driver->ops->shutdown(substream, codec_dai);
-	}
+	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai)
+		snd_soc_dai_shutdown(codec_dai, substream);
 
 component_err:
 	soc_pcm_components_close(substream, component);
 
-	if (cpu_dai->driver->ops->shutdown)
-		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+	snd_soc_dai_shutdown(cpu_dai, substream);
 out:
 	mutex_unlock(&rtd->pcm_mutex);
 
@@ -728,13 +725,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 
 	snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream);
 
-	if (cpu_dai->driver->ops->shutdown)
-		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+	snd_soc_dai_shutdown(cpu_dai, substream);
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->driver->ops->shutdown)
-			codec_dai->driver->ops->shutdown(substream, codec_dai);
-	}
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		snd_soc_dai_shutdown(codec_dai, substream);
 
 	if (rtd->dai_link->ops->shutdown)
 		rtd->dai_link->ops->shutdown(substream);
-- 
cgit v1.2.3-59-g8ed1b


From 4beb8e109d30d339d44308a767dd6f5614492f3e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:33:45 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_prepare()

Current ALSA SoC is directly using dai->driver->ops->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_prepare() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tvbehn46.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-dai.c     | 11 +++++++++++
 sound/soc/soc-pcm.c     | 27 +++++++++++----------------
 3 files changed, 24 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 32545d457b3d..c7dff6a0b5b9 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -154,6 +154,8 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream);
 void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			  struct snd_pcm_substream *substream);
+int snd_soc_dai_prepare(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 67ff6cc1fe02..cb810888c563 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -307,3 +307,14 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 	if (dai->driver->ops->shutdown)
 		dai->driver->ops->shutdown(substream, dai);
 }
+
+int snd_soc_dai_prepare(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->prepare)
+		ret = dai->driver->ops->prepare(substream, dai);
+
+	return ret;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index ed5ae23c7104..d7611af90dce 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -814,27 +814,22 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	}
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->driver->ops->prepare) {
-			ret = codec_dai->driver->ops->prepare(substream,
-							      codec_dai);
-			if (ret < 0) {
-				dev_err(codec_dai->dev,
-					"ASoC: codec DAI prepare error: %d\n",
-					ret);
-				goto out;
-			}
-		}
-	}
-
-	if (cpu_dai->driver->ops->prepare) {
-		ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
+		ret = snd_soc_dai_prepare(codec_dai, substream);
 		if (ret < 0) {
-			dev_err(cpu_dai->dev,
-				"ASoC: cpu DAI prepare error: %d\n", ret);
+			dev_err(codec_dai->dev,
+				"ASoC: codec DAI prepare error: %d\n",
+				ret);
 			goto out;
 		}
 	}
 
+	ret = snd_soc_dai_prepare(cpu_dai, substream);
+	if (ret < 0) {
+		dev_err(cpu_dai->dev,
+			"ASoC: cpu DAI prepare error: %d\n", ret);
+		goto out;
+	}
+
 	/* cancel any delayed stream shutdown that is pending */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
 	    rtd->pop_wait) {
-- 
cgit v1.2.3-59-g8ed1b


From 95aef35533844f35544851b0cdc1fc154b603307 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:33:51 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_trigger()

Current ALSA SoC is directly using dai->driver->ops->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_trigger() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgqyhn40.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-dai.c     | 12 ++++++++++++
 sound/soc/soc-pcm.c     | 17 ++++++-----------
 3 files changed, 20 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index c7dff6a0b5b9..72b8e76f1cc4 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -156,6 +156,8 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			  struct snd_pcm_substream *substream);
 int snd_soc_dai_prepare(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream);
+int snd_soc_dai_trigger(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream, int cmd);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index cb810888c563..18c447e169f6 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -318,3 +318,15 @@ int snd_soc_dai_prepare(struct snd_soc_dai *dai,
 
 	return ret;
 }
+
+int snd_soc_dai_trigger(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream,
+			int cmd)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->trigger)
+		ret = dai->driver->ops->trigger(substream, cmd, dai);
+
+	return ret;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d7611af90dce..a628b08f966e 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1084,12 +1084,9 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	int i, ret;
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->driver->ops->trigger) {
-			ret = codec_dai->driver->ops->trigger(substream,
-							      cmd, codec_dai);
-			if (ret < 0)
-				return ret;
-		}
+		ret = snd_soc_dai_trigger(codec_dai, substream, cmd);
+		if (ret < 0)
+			return ret;
 	}
 
 	for_each_rtdcom(rtd, rtdcom) {
@@ -1104,11 +1101,9 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 			return ret;
 	}
 
-	if (cpu_dai->driver->ops->trigger) {
-		ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
-		if (ret < 0)
-			return ret;
-	}
+	snd_soc_dai_trigger(cpu_dai, substream, cmd);
+	if (ret < 0)
+		return ret;
 
 	if (rtd->dai_link->ops->trigger) {
 		ret = rtd->dai_link->ops->trigger(substream, cmd);
-- 
cgit v1.2.3-59-g8ed1b


From 5c0769af4caf8fbdad2e9c0051ab0081b8e22b0a Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:33:56 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_bespoke_trigger()

Current ALSA SoC is directly using dai->driver->ops->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_bespoke_trigger() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r26ihn3u.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-dai.c     | 12 ++++++++++++
 sound/soc/soc-pcm.c     | 16 ++++++----------
 3 files changed, 20 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 72b8e76f1cc4..6a5566d459ad 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -158,6 +158,8 @@ int snd_soc_dai_prepare(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream);
 int snd_soc_dai_trigger(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream, int cmd);
+int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream, int cmd);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 18c447e169f6..6f466cfcbeef 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -330,3 +330,15 @@ int snd_soc_dai_trigger(struct snd_soc_dai *dai,
 
 	return ret;
 }
+
+int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
+				struct snd_pcm_substream *substream,
+				int cmd)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->bespoke_trigger)
+		ret = dai->driver->ops->bespoke_trigger(substream, cmd, dai);
+
+	return ret;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index a628b08f966e..a10627f1ceff 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1123,19 +1123,15 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
 	int i, ret;
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->driver->ops->bespoke_trigger) {
-			ret = codec_dai->driver->ops->bespoke_trigger(substream,
-								cmd, codec_dai);
-			if (ret < 0)
-				return ret;
-		}
-	}
-
-	if (cpu_dai->driver->ops->bespoke_trigger) {
-		ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai);
+		ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd);
 		if (ret < 0)
 			return ret;
 	}
+
+	snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
+	if (ret < 0)
+		return ret;
+
 	return 0;
 }
 /*
-- 
cgit v1.2.3-59-g8ed1b


From 1dea80d4b2bd3b53c58f008ca2bcd73182583711 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:34:09 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_delay()

Current ALSA SoC is directly using dai->driver->ops->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_delay() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o91mhn3i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-dai.c     | 11 +++++++++++
 sound/soc/soc-pcm.c     |  9 +++------
 3 files changed, 16 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 6a5566d459ad..7cfed3034511 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -160,6 +160,8 @@ int snd_soc_dai_trigger(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream, int cmd);
 int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream, int cmd);
+snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
+				    struct snd_pcm_substream *substream);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 6f466cfcbeef..5b5b979cd1f3 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -342,3 +342,14 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
 
 	return ret;
 }
+
+snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
+				    struct snd_pcm_substream *substream)
+{
+	int delay = 0;
+
+	if (dai->driver->ops->delay)
+		delay = dai->driver->ops->delay(substream, dai);
+
+	return delay;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index a10627f1ceff..f3137723301c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1169,14 +1169,11 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	/* base delay if assigned in pointer callback */
 	delay = runtime->delay;
 
-	if (cpu_dai->driver->ops->delay)
-		delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
+	delay += snd_soc_dai_delay(cpu_dai, substream);
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->driver->ops->delay)
-			codec_delay = max(codec_delay,
-					codec_dai->driver->ops->delay(substream,
-								    codec_dai));
+		codec_delay = max(codec_delay,
+				  snd_soc_dai_delay(codec_dai, substream));
 	}
 	delay += codec_delay;
 
-- 
cgit v1.2.3-59-g8ed1b


From e0f2262292d0c8160cfd9a8c40425107fb65ab29 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:34:29 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_suspend()

Current ALSA SoC is directly using dai->driver->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_suspend() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87muh6hn2x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h | 1 +
 sound/soc/soc-core.c    | 8 ++++----
 sound/soc/soc-dai.c     | 6 ++++++
 3 files changed, 11 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 7cfed3034511..6c5604a7dbc2 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -162,6 +162,7 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream, int cmd);
 snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
 				    struct snd_pcm_substream *substream);
+void snd_soc_dai_suspend(struct snd_soc_dai *dai);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6e8c5c8eeaec..7493afb2371c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -511,8 +511,8 @@ int snd_soc_suspend(struct device *dev)
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		if (cpu_dai->driver->suspend && !cpu_dai->driver->bus_control)
-			cpu_dai->driver->suspend(cpu_dai);
+		if (!cpu_dai->driver->bus_control)
+			snd_soc_dai_suspend(cpu_dai);
 	}
 
 	/* close any waiting streams */
@@ -584,8 +584,8 @@ int snd_soc_suspend(struct device *dev)
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		if (cpu_dai->driver->suspend && cpu_dai->driver->bus_control)
-			cpu_dai->driver->suspend(cpu_dai);
+		if (cpu_dai->driver->bus_control)
+			snd_soc_dai_suspend(cpu_dai);
 
 		/* deactivate pins to sleep state */
 		pinctrl_pm_select_sleep_state(cpu_dai->dev);
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 5b5b979cd1f3..3373598e0682 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -353,3 +353,9 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
 
 	return delay;
 }
+
+void snd_soc_dai_suspend(struct snd_soc_dai *dai)
+{
+	if (dai->driver->suspend)
+		dai->driver->suspend(dai);
+}
-- 
cgit v1.2.3-59-g8ed1b


From 24b09d051164680f0a1d1910efe21ce36ad5c1ca Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:34:43 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_resume()

Current ALSA SoC is directly using dai->driver->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_resume() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lfwqhn2j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h | 1 +
 sound/soc/soc-core.c    | 8 ++++----
 sound/soc/soc-dai.c     | 6 ++++++
 3 files changed, 11 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 6c5604a7dbc2..ed78e34a814e 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -163,6 +163,7 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
 snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
 				    struct snd_pcm_substream *substream);
 void snd_soc_dai_suspend(struct snd_soc_dai *dai);
+void snd_soc_dai_resume(struct snd_soc_dai *dai);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7493afb2371c..5c02f90cea69 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -631,8 +631,8 @@ static void soc_resume_deferred(struct work_struct *work)
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		if (cpu_dai->driver->resume && cpu_dai->driver->bus_control)
-			cpu_dai->driver->resume(cpu_dai);
+		if (cpu_dai->driver->bus_control)
+			snd_soc_dai_resume(cpu_dai);
 	}
 
 	for_each_card_components(card, component) {
@@ -678,8 +678,8 @@ static void soc_resume_deferred(struct work_struct *work)
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		if (cpu_dai->driver->resume && !cpu_dai->driver->bus_control)
-			cpu_dai->driver->resume(cpu_dai);
+		if (!cpu_dai->driver->bus_control)
+			snd_soc_dai_resume(cpu_dai);
 	}
 
 	if (card->resume_post)
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 3373598e0682..ddb6f217c0ed 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -359,3 +359,9 @@ void snd_soc_dai_suspend(struct snd_soc_dai *dai)
 	if (dai->driver->suspend)
 		dai->driver->suspend(dai);
 }
+
+void snd_soc_dai_resume(struct snd_soc_dai *dai)
+{
+	if (dai->driver->resume)
+		dai->driver->resume(dai);
+}
-- 
cgit v1.2.3-59-g8ed1b


From cfd9b5fbfe1e8763018aea2600aa0d6ff015ebfc Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:34:56 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_probe()

Current ALSA SoC is directly using dai->driver->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_probe() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k1cahn26.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  1 +
 sound/soc/soc-core.c    | 15 +++++++--------
 sound/soc/soc-dai.c     |  7 +++++++
 3 files changed, 15 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index ed78e34a814e..da8d8b889089 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -164,6 +164,7 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
 				    struct snd_pcm_substream *substream);
 void snd_soc_dai_suspend(struct snd_soc_dai *dai);
 void snd_soc_dai_resume(struct snd_soc_dai *dai);
+int snd_soc_dai_probe(struct snd_soc_dai *dai);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 5c02f90cea69..3e73468225f9 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1434,18 +1434,17 @@ static int soc_probe_link_components(struct snd_soc_card *card,
 
 static int soc_probe_dai(struct snd_soc_dai *dai, int order)
 {
+	int ret;
+
 	if (dai->probed ||
 	    dai->driver->probe_order != order)
 		return 0;
 
-	if (dai->driver->probe) {
-		int ret = dai->driver->probe(dai);
-
-		if (ret < 0) {
-			dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
-				dai->name, ret);
-			return ret;
-		}
+	ret = snd_soc_dai_probe(dai);
+	if (ret < 0) {
+		dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
+			dai->name, ret);
+		return ret;
 	}
 
 	dai->probed = 1;
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index ddb6f217c0ed..55c1fac99613 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -365,3 +365,10 @@ void snd_soc_dai_resume(struct snd_soc_dai *dai)
 	if (dai->driver->resume)
 		dai->driver->resume(dai);
 }
+
+int snd_soc_dai_probe(struct snd_soc_dai *dai)
+{
+	if (dai->driver->probe)
+		return dai->driver->probe(dai);
+	return 0;
+}
-- 
cgit v1.2.3-59-g8ed1b


From dcdab5820edd6123911dbd767ee1e389008b6a83 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:35:05 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_remove()

Current ALSA SoC is directly using dai->driver->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_remvoe() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87imruhn1x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  1 +
 sound/soc/soc-core.c    | 13 ++++++-------
 sound/soc/soc-dai.c     |  7 +++++++
 3 files changed, 14 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index da8d8b889089..2a11f177ce01 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -165,6 +165,7 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
 void snd_soc_dai_suspend(struct snd_soc_dai *dai);
 void snd_soc_dai_resume(struct snd_soc_dai *dai);
 int snd_soc_dai_probe(struct snd_soc_dai *dai);
+int snd_soc_dai_remove(struct snd_soc_dai *dai);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3e73468225f9..727fd342b3fb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -992,13 +992,12 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 	    dai->driver->remove_order != order)
 		return;
 
-	if (dai->driver->remove) {
-		err = dai->driver->remove(dai);
-		if (err < 0)
-			dev_err(dai->dev,
-				"ASoC: failed to remove %s: %d\n",
-				dai->name, err);
-	}
+	err = snd_soc_dai_remove(dai);
+	if (err < 0)
+		dev_err(dai->dev,
+			"ASoC: failed to remove %s: %d\n",
+			dai->name, err);
+
 	dai->probed = 0;
 }
 
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 55c1fac99613..384765c747da 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -372,3 +372,10 @@ int snd_soc_dai_probe(struct snd_soc_dai *dai)
 		return dai->driver->probe(dai);
 	return 0;
 }
+
+int snd_soc_dai_remove(struct snd_soc_dai *dai)
+{
+	if (dai->driver->remove)
+		return dai->driver->remove(dai);
+	return 0;
+}
-- 
cgit v1.2.3-59-g8ed1b


From b423c4202135f7794e0a9c55a884f5933d8e7156 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:35:29 +0900
Subject: ASoC: soc-dai: add snd_soc_dai_compress_new()

Current ALSA SoC is directly using dai->driver->xxx,
thus, it has deep nested bracket, and it makes code unreadable.
This patch adds new snd_soc_dai_compress_new() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h87ehn1a.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h |  2 ++
 sound/soc/soc-core.c    | 15 ++++++++-------
 sound/soc/soc-dai.c     |  8 ++++++++
 3 files changed, 18 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 2a11f177ce01..0f8b09520020 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -166,6 +166,8 @@ void snd_soc_dai_suspend(struct snd_soc_dai *dai);
 void snd_soc_dai_resume(struct snd_soc_dai *dai);
 int snd_soc_dai_probe(struct snd_soc_dai *dai);
 int snd_soc_dai_remove(struct snd_soc_dai *dai);
+int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
+			     struct snd_soc_pcm_runtime *rtd, int num);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 727fd342b3fb..458b090f026a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1545,15 +1545,16 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 			num = rtd->dai_link->id;
 	}
 
-	if (cpu_dai->driver->compress_new) {
-		/* create compress_device" */
-		ret = cpu_dai->driver->compress_new(rtd, num);
-		if (ret < 0) {
+	/* create compress_device if possible */
+	ret = snd_soc_dai_compress_new(cpu_dai, rtd, num);
+	if (ret != -ENOTSUPP) {
+		if (ret < 0)
 			dev_err(card->dev, "ASoC: can't create compress %s\n",
 					 dai_link->stream_name);
-			return ret;
-		}
-	} else if (!dai_link->params) {
+		return ret;
+	}
+
+	if (!dai_link->params) {
 		/* create the pcm */
 		ret = soc_new_pcm(rtd, num);
 		if (ret < 0) {
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 384765c747da..e6f161b9f975 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -379,3 +379,11 @@ int snd_soc_dai_remove(struct snd_soc_dai *dai)
 		return dai->driver->remove(dai);
 	return 0;
 }
+
+int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
+			     struct snd_soc_pcm_runtime *rtd, int num)
+{
+	if (dai->driver->compress_new)
+		return dai->driver->compress_new(rtd, num);
+	return -ENOTSUPP;
+}
-- 
cgit v1.2.3-59-g8ed1b


From 88fdffa2eef45fc9f6009ee590741c44a57c49c7 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:36:27 +0900
Subject: ASoC: soc-core: use existing snd_soc_dai_digital_mute()

ALSA SoC already has snd_soc_dai_digital_mute() for digital_mute,
let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ef2ihmzo.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 14 ++++++--------
 1 file changed, 6 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 458b090f026a..7ecfe641ca46 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -487,10 +487,9 @@ int snd_soc_suspend(struct device *dev)
 			continue;
 
 		for_each_rtd_codec_dai(rtd, i, dai) {
-			struct snd_soc_dai_driver *drv = dai->driver;
-
-			if (drv->ops->digital_mute && dai->playback_active)
-				drv->ops->digital_mute(dai, 1);
+			if (dai->playback_active)
+				snd_soc_dai_digital_mute(dai, 1,
+						SNDRV_PCM_STREAM_PLAYBACK);
 		}
 	}
 
@@ -665,10 +664,9 @@ static void soc_resume_deferred(struct work_struct *work)
 			continue;
 
 		for_each_rtd_codec_dai(rtd, i, dai) {
-			struct snd_soc_dai_driver *drv = dai->driver;
-
-			if (drv->ops->digital_mute && dai->playback_active)
-				drv->ops->digital_mute(dai, 0);
+			if (dai->playback_active)
+				snd_soc_dai_digital_mute(dai, 0,
+						SNDRV_PCM_STREAM_PLAYBACK);
 		}
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From 467fece8fbc6774a3a3bd0981e1a342fb5022706 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 22 Jul 2019 10:36:16 +0900
Subject: ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c

snd_soc_dai_stream_valid() is function to check stream validity.
But, some code is using it, some code are checking stream->channels_min
directly. Doing samethings by different method is confusable.
This patch uses same funcntion for same purpose.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ftmyhmzz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dai.h  |  1 +
 sound/soc/soc-compress.c |  9 ++++-----
 sound/soc/soc-dai.c      | 18 ++++++++++++++++++
 sound/soc/soc-pcm.c      | 39 ++++++++++-----------------------------
 4 files changed, 33 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 0f8b09520020..dc48fe081a20 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -168,6 +168,7 @@ int snd_soc_dai_probe(struct snd_soc_dai *dai);
 int snd_soc_dai_remove(struct snd_soc_dai *dai);
 int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
 			     struct snd_soc_pcm_runtime *rtd, int num);
+bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
 
 struct snd_soc_dai_ops {
 	/*
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index ddef4ff677ce..289211069a1e 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -872,14 +872,13 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 	}
 
 	/* check client and interface hw capabilities */
-	if (codec_dai->driver->playback.channels_min)
+	if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
+	    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK))
 		playback = 1;
-	if (codec_dai->driver->capture.channels_min)
+	if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) &&
+	    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_CAPTURE))
 		capture = 1;
 
-	capture = capture && cpu_dai->driver->capture.channels_min;
-	playback = playback && cpu_dai->driver->playback.channels_min;
-
 	/*
 	 * Compress devices are unidirectional so only one of the directions
 	 * should be set, check for that (xor)
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index e6f161b9f975..1c7f63871c1d 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -387,3 +387,21 @@ int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
 		return dai->driver->compress_new(rtd, num);
 	return -ENOTSUPP;
 }
+
+/*
+ * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
+ *
+ * Returns true if the DAI supports the indicated stream type.
+ */
+bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
+{
+	struct snd_soc_pcm_stream *stream;
+
+	if (dir == SNDRV_PCM_STREAM_PLAYBACK)
+		stream = &dai->driver->playback;
+	else
+		stream = &dai->driver->capture;
+
+	/* If the codec specifies any channels at all, it supports the stream */
+	return stream->channels_min;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index f3137723301c..fabeac164a6c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -29,24 +29,6 @@
 
 #define DPCM_MAX_BE_USERS	8
 
-/*
- * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
- *
- * Returns true if the DAI supports the indicated stream type.
- */
-static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
-{
-	struct snd_soc_pcm_stream *codec_stream;
-
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-		codec_stream = &dai->driver->playback;
-	else
-		codec_stream = &dai->driver->capture;
-
-	/* If the codec specifies any channels at all, it supports the stream */
-	return codec_stream->channels_min;
-}
-
 /**
  * snd_soc_runtime_activate() - Increment active count for PCM runtime components
  * @rtd: ASoC PCM runtime that is activated
@@ -2688,8 +2670,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 		new ? "new" : "old", fe->dai_link->name);
 
 	/* skip if FE doesn't have playback capability */
-	if (!fe->cpu_dai->driver->playback.channels_min ||
-	    !fe->codec_dai->driver->playback.channels_min)
+	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK) ||
+	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK))
 		goto capture;
 
 	/* skip if FE isn't currently playing */
@@ -2719,8 +2701,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 
 capture:
 	/* skip if FE doesn't have capture capability */
-	if (!fe->cpu_dai->driver->capture.channels_min ||
-	    !fe->codec_dai->driver->capture.channels_min)
+	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_CAPTURE) ||
+	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE))
 		return 0;
 
 	/* skip if FE isn't currently capturing */
@@ -3030,14 +3012,13 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
 		for_each_rtd_codec_dai(rtd, i, codec_dai) {
-			if (codec_dai->driver->playback.channels_min)
+			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
+			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK))
 				playback = 1;
-			if (codec_dai->driver->capture.channels_min)
+			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) &&
+			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_CAPTURE))
 				capture = 1;
 		}
-
-		capture = capture && cpu_dai->driver->capture.channels_min;
-		playback = playback && cpu_dai->driver->playback.channels_min;
 	}
 
 	if (rtd->dai_link->playback_only) {
@@ -3375,11 +3356,11 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
-	if (fe->cpu_dai->driver->playback.channels_min)
+	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
 		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
 					buf + offset, out_count - offset);
 
-	if (fe->cpu_dai->driver->capture.channels_min)
+	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE))
 		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
 					buf + offset, out_count - offset);
 
-- 
cgit v1.2.3-59-g8ed1b


From a30f1743e4f57b359903fd679476b2903e7e5dac Mon Sep 17 00:00:00 2001
From: Pavel Machek <pavel@denx.de>
Date: Wed, 24 Jul 2019 11:52:54 +0200
Subject: ALSA: line6: sizeof (byte) is always 1, use that fact.

sizeof (byte) is always 1, use that fact and make interesting code explicit.

Signed-off-by: Pavel Machek <pavel@denx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/line6/driver.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index ab2ec896f49c..b5a3f754a4f1 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -342,7 +342,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data,
 	if (address > 0xffff || datalen > 0xff)
 		return -EINVAL;
 
-	len = kmalloc(sizeof(*len), GFP_KERNEL);
+	len = kmalloc(1, GFP_KERNEL);
 	if (!len)
 		return -ENOMEM;
 
@@ -418,7 +418,7 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data,
 	if (address > 0xffff || datalen > 0xffff)
 		return -EINVAL;
 
-	status = kmalloc(sizeof(*status), GFP_KERNEL);
+	status = kmalloc(1, GFP_KERNEL);
 	if (!status)
 		return -ENOMEM;
 
-- 
cgit v1.2.3-59-g8ed1b


From e8758a5ed2783c417be1f5aab5af9fe4be60956f Mon Sep 17 00:00:00 2001
From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Date: Wed, 24 Jul 2019 07:26:32 +0200
Subject: ASoC: wm8955: Fix a typo in 'wm8995_pll_factors()' function name

This should be 'wm8955_pll_factors()' instead.
Fix it and use it.

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190724052632.30476-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8955.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 66a5f1827aa9..9c7e2892c8cb 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -140,7 +140,7 @@ struct pll_factors {
  * to allow rounding later */
 #define FIXED_FLL_SIZE ((1 << 22) * 10)
 
-static int wm8995_pll_factors(struct device *dev,
+static int wm8955_pll_factors(struct device *dev,
 			      int Fref, int Fout, struct pll_factors *pll)
 {
 	u64 Kpart;
@@ -279,7 +279,7 @@ static int wm8955_configure_clocking(struct snd_soc_component *component)
 
 		/* Use the last divider configuration we saw for the
 		 * sample rate. */
-		ret = wm8995_pll_factors(component->dev, wm8955->mclk_rate,
+		ret = wm8955_pll_factors(component->dev, wm8955->mclk_rate,
 					 clock_cfgs[sr].mclk, &pll);
 		if (ret != 0) {
 			dev_err(component->dev,
-- 
cgit v1.2.3-59-g8ed1b


From bcc2a2dc3ba8c3a7aed856f840afa6a47e3cb8e0 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 23 Jul 2019 16:58:48 +0200
Subject: ASoC: Intel: Skylake: Merge skl_sst and skl into skl_dev struct

Skylake driver is divided into two modules:
- snd_soc_skl
- snd_soc_skl_ipc

and nothing would be wrong if not for the fact that both cannot exist
without one another. IPC module is not some kind of extension, as it is
the case for snd_hda_ext_core which is separated from snd_hda_core -
legacy hda interface. It's as much core Skylake module as snd_soc_skl
is.

Statement backed up by existence of circular dependency between this
two. To eliminate said problem, struct skl_sst has been created. From
that very momment, Skylake has been plagued by header errors (incomplete
structs, unknown references etc.) whenever something new is to be added
or code is cleaned up.

As this design is being corrected, struct skl_sst is no longer needed,
so combine it with struct skl. To avoid ambiguity when searching for skl
stuff (struct skl *skl) it has also been renamed to skl_dev.

No functional changes.

Signed-off-by: Piotr Maziarz <piotrx.maziarz@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723145854.8527-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/common/sst-ipc.h        |   1 +
 sound/soc/intel/skylake/bxt-sst.c       |  50 ++---
 sound/soc/intel/skylake/cnl-sst-dsp.h   |   7 +-
 sound/soc/intel/skylake/cnl-sst.c       |  37 ++--
 sound/soc/intel/skylake/skl-debug.c     |   8 +-
 sound/soc/intel/skylake/skl-messages.c  | 229 +++++++++++------------
 sound/soc/intel/skylake/skl-nhlt.c      |  18 +-
 sound/soc/intel/skylake/skl-pcm.c       |  75 ++++----
 sound/soc/intel/skylake/skl-ssp-clk.c   |   4 +-
 sound/soc/intel/skylake/skl-sst-dsp.c   |  10 +-
 sound/soc/intel/skylake/skl-sst-dsp.h   |  30 +--
 sound/soc/intel/skylake/skl-sst-ipc.c   |   8 +-
 sound/soc/intel/skylake/skl-sst-ipc.h   |  52 +-----
 sound/soc/intel/skylake/skl-sst-utils.c |  37 ++--
 sound/soc/intel/skylake/skl-sst.c       |  51 ++---
 sound/soc/intel/skylake/skl-topology.c  | 320 +++++++++++++++-----------------
 sound/soc/intel/skylake/skl-topology.h  |  32 ++--
 sound/soc/intel/skylake/skl.c           |  54 +++---
 sound/soc/intel/skylake/skl.h           |  94 +++++++---
 19 files changed, 545 insertions(+), 572 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h
index c6779e2ac830..ef38600e88f7 100644
--- a/sound/soc/intel/common/sst-ipc.h
+++ b/sound/soc/intel/common/sst-ipc.h
@@ -35,6 +35,7 @@ struct ipc_message {
 };
 
 struct sst_generic_ipc;
+struct sst_dsp;
 
 struct sst_plat_ipc_ops {
 	void (*tx_msg)(struct sst_generic_ipc *, struct ipc_message *);
diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index 46d5159cf905..92a82e6b5fe6 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -14,7 +14,7 @@
 
 #include "../common/sst-dsp.h"
 #include "../common/sst-dsp-priv.h"
-#include "skl-sst-ipc.h"
+#include "skl.h"
 
 #define BXT_BASEFW_TIMEOUT	3000
 #define BXT_INIT_TIMEOUT	300
@@ -49,7 +49,7 @@ static int
 bxt_load_library(struct sst_dsp *ctx, struct skl_lib_info *linfo, int lib_count)
 {
 	struct snd_dma_buffer dmab;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	struct firmware stripped_fw;
 	int ret = 0, i, dma_id, stream_tag;
 
@@ -184,7 +184,7 @@ static int sst_transfer_fw_host_dma(struct sst_dsp *ctx)
 static int bxt_load_base_firmware(struct sst_dsp *ctx)
 {
 	struct firmware stripped_fw;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	int ret, i;
 
 	if (ctx->fw == NULL) {
@@ -268,7 +268,7 @@ sst_load_base_firmware_failed:
  */
 static int bxt_d0i3_target_state(struct sst_dsp *ctx)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	struct skl_d0i3_data *d0i3 = &skl->d0i3;
 
 	if (skl->cores.state[SKL_DSP_CORE0_ID] != SKL_DSP_RUNNING)
@@ -288,8 +288,8 @@ static void bxt_set_dsp_D0i3(struct work_struct *work)
 {
 	int ret;
 	struct skl_ipc_d0ix_msg msg;
-	struct skl_sst *skl = container_of(work,
-			struct skl_sst, d0i3.work.work);
+	struct skl_dev *skl = container_of(work,
+			struct skl_dev, d0i3.work.work);
 	struct sst_dsp *ctx = skl->dsp;
 	struct skl_d0i3_data *d0i3 = &skl->d0i3;
 	int target_state;
@@ -331,7 +331,7 @@ static void bxt_set_dsp_D0i3(struct work_struct *work)
 
 static int bxt_schedule_dsp_D0i3(struct sst_dsp *ctx)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	struct skl_d0i3_data *d0i3 = &skl->d0i3;
 
 	/* Schedule D0i3 only if the usecase ref counts are appropriate */
@@ -350,7 +350,7 @@ static int bxt_set_dsp_D0i0(struct sst_dsp *ctx)
 {
 	int ret;
 	struct skl_ipc_d0ix_msg msg;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 
 	dev_dbg(ctx->dev, "In %s:\n", __func__);
 
@@ -389,7 +389,7 @@ static int bxt_set_dsp_D0i0(struct sst_dsp *ctx)
 
 static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	int ret;
 	struct skl_ipc_dxstate_info dx;
 	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
@@ -486,7 +486,7 @@ static int bxt_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id)
 {
 	int ret;
 	struct skl_ipc_dxstate_info dx;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
 
 	dx.core_mask = core_mask;
@@ -548,9 +548,9 @@ static struct sst_dsp_device skl_dev = {
 
 int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 			const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
-			struct skl_sst **dsp)
+			struct skl_dev **dsp)
 {
-	struct skl_sst *skl;
+	struct skl_dev *skl;
 	struct sst_dsp *sst;
 	int ret;
 
@@ -591,10 +591,10 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 }
 EXPORT_SYMBOL_GPL(bxt_sst_dsp_init);
 
-int bxt_sst_init_fw(struct device *dev, struct skl_sst *ctx)
+int bxt_sst_init_fw(struct device *dev, struct skl_dev *skl)
 {
 	int ret;
-	struct sst_dsp *sst = ctx->dsp;
+	struct sst_dsp *sst = skl->dsp;
 
 	ret = sst->fw_ops.load_fw(sst);
 	if (ret < 0) {
@@ -604,29 +604,29 @@ int bxt_sst_init_fw(struct device *dev, struct skl_sst *ctx)
 
 	skl_dsp_init_core_state(sst);
 
-	if (ctx->lib_count > 1) {
-		ret = sst->fw_ops.load_library(sst, ctx->lib_info,
-						ctx->lib_count);
+	if (skl->lib_count > 1) {
+		ret = sst->fw_ops.load_library(sst, skl->lib_info,
+						skl->lib_count);
 		if (ret < 0) {
 			dev_err(dev, "Load Library failed : %x\n", ret);
 			return ret;
 		}
 	}
-	ctx->is_first_boot = false;
+	skl->is_first_boot = false;
 
 	return 0;
 }
 EXPORT_SYMBOL_GPL(bxt_sst_init_fw);
 
-void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl)
 {
 
-	skl_release_library(ctx->lib_info, ctx->lib_count);
-	if (ctx->dsp->fw)
-		release_firmware(ctx->dsp->fw);
-	skl_freeup_uuid_list(ctx);
-	skl_ipc_free(&ctx->ipc);
-	ctx->dsp->ops->free(ctx->dsp);
+	skl_release_library(skl->lib_info, skl->lib_count);
+	if (skl->dsp->fw)
+		release_firmware(skl->dsp->fw);
+	skl_freeup_uuid_list(skl);
+	skl_ipc_free(&skl->ipc);
+	skl->dsp->ops->free(skl->dsp);
 }
 EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup);
 
diff --git a/sound/soc/intel/skylake/cnl-sst-dsp.h b/sound/soc/intel/skylake/cnl-sst-dsp.h
index 426515faab52..7bd4d2a8fdfa 100644
--- a/sound/soc/intel/skylake/cnl-sst-dsp.h
+++ b/sound/soc/intel/skylake/cnl-sst-dsp.h
@@ -9,7 +9,6 @@
 #define __CNL_SST_DSP_H__
 
 struct sst_dsp;
-struct skl_sst;
 struct sst_dsp_device;
 struct sst_generic_ipc;
 
@@ -97,8 +96,8 @@ void cnl_ipc_free(struct sst_generic_ipc *ipc);
 
 int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 		     const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
-		     struct skl_sst **dsp);
-int cnl_sst_init_fw(struct device *dev, struct skl_sst *ctx);
-void cnl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
+		     struct skl_dev **dsp);
+int cnl_sst_init_fw(struct device *dev, struct skl_dev *skl);
+void cnl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl);
 
 #endif /*__CNL_SST_DSP_H__*/
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
index f2c09fa6ea40..2d748a335bcf 100644
--- a/sound/soc/intel/skylake/cnl-sst.c
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -24,8 +24,7 @@
 #include "../common/sst-dsp-priv.h"
 #include "../common/sst-ipc.h"
 #include "cnl-sst-dsp.h"
-#include "skl-sst-dsp.h"
-#include "skl-sst-ipc.h"
+#include "skl.h"
 
 #define CNL_FW_ROM_INIT		0x1
 #define CNL_FW_INIT		0x5
@@ -109,7 +108,7 @@ static int sst_transfer_fw_host_dma(struct sst_dsp *ctx)
 static int cnl_load_base_firmware(struct sst_dsp *ctx)
 {
 	struct firmware stripped_fw;
-	struct skl_sst *cnl = ctx->thread_context;
+	struct skl_dev *cnl = ctx->thread_context;
 	int ret;
 
 	if (!ctx->fw) {
@@ -167,7 +166,7 @@ cnl_load_base_firmware_failed:
 
 static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id)
 {
-	struct skl_sst *cnl = ctx->thread_context;
+	struct skl_dev *cnl = ctx->thread_context;
 	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
 	struct skl_ipc_dxstate_info dx;
 	int ret;
@@ -229,7 +228,7 @@ err:
 
 static int cnl_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id)
 {
-	struct skl_sst *cnl = ctx->thread_context;
+	struct skl_dev *cnl = ctx->thread_context;
 	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
 	struct skl_ipc_dxstate_info dx;
 	int ret;
@@ -293,7 +292,7 @@ static struct sst_ops cnl_ops = {
 static irqreturn_t cnl_dsp_irq_thread_handler(int irq, void *context)
 {
 	struct sst_dsp *dsp = context;
-	struct skl_sst *cnl = sst_dsp_get_thread_context(dsp);
+	struct skl_dev *cnl = sst_dsp_get_thread_context(dsp);
 	struct sst_generic_ipc *ipc = &cnl->ipc;
 	struct skl_ipc_header header = {0};
 	u32 hipcida, hipctdr, hipctdd;
@@ -386,7 +385,7 @@ static bool cnl_ipc_is_dsp_busy(struct sst_dsp *dsp)
 	return (hipcidr & CNL_ADSP_REG_HIPCIDR_BUSY);
 }
 
-static int cnl_ipc_init(struct device *dev, struct skl_sst *cnl)
+static int cnl_ipc_init(struct device *dev, struct skl_dev *cnl)
 {
 	struct sst_generic_ipc *ipc;
 	int err;
@@ -415,9 +414,9 @@ static int cnl_ipc_init(struct device *dev, struct skl_sst *cnl)
 
 int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 		     const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
-		     struct skl_sst **dsp)
+		     struct skl_dev **dsp)
 {
-	struct skl_sst *cnl;
+	struct skl_dev *cnl;
 	struct sst_dsp *sst;
 	int ret;
 
@@ -454,12 +453,12 @@ int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 }
 EXPORT_SYMBOL_GPL(cnl_sst_dsp_init);
 
-int cnl_sst_init_fw(struct device *dev, struct skl_sst *ctx)
+int cnl_sst_init_fw(struct device *dev, struct skl_dev *skl)
 {
 	int ret;
-	struct sst_dsp *sst = ctx->dsp;
+	struct sst_dsp *sst = skl->dsp;
 
-	ret = ctx->dsp->fw_ops.load_fw(sst);
+	ret = skl->dsp->fw_ops.load_fw(sst);
 	if (ret < 0) {
 		dev_err(dev, "load base fw failed: %d", ret);
 		return ret;
@@ -467,21 +466,21 @@ int cnl_sst_init_fw(struct device *dev, struct skl_sst *ctx)
 
 	skl_dsp_init_core_state(sst);
 
-	ctx->is_first_boot = false;
+	skl->is_first_boot = false;
 
 	return 0;
 }
 EXPORT_SYMBOL_GPL(cnl_sst_init_fw);
 
-void cnl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
+void cnl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl)
 {
-	if (ctx->dsp->fw)
-		release_firmware(ctx->dsp->fw);
+	if (skl->dsp->fw)
+		release_firmware(skl->dsp->fw);
 
-	skl_freeup_uuid_list(ctx);
-	cnl_ipc_free(&ctx->ipc);
+	skl_freeup_uuid_list(skl);
+	cnl_ipc_free(&skl->ipc);
 
-	ctx->dsp->ops->free(ctx->dsp);
+	skl->dsp->ops->free(skl->dsp);
 }
 EXPORT_SYMBOL_GPL(cnl_sst_dsp_cleanup);
 
diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index b9b4a72a4334..c43aa4081232 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -20,7 +20,7 @@
 #define FW_REG_SIZE	0x60
 
 struct skl_debug {
-	struct skl *skl;
+	struct skl_dev *skl;
 	struct device *dev;
 
 	struct dentry *fs;
@@ -172,7 +172,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf,
 			       size_t count, loff_t *ppos)
 {
 	struct skl_debug *d = file->private_data;
-	struct sst_dsp *sst = d->skl->skl_sst->dsp;
+	struct sst_dsp *sst = d->skl->dsp;
 	size_t w0_stat_sz = sst->addr.w0_stat_sz;
 	void __iomem *in_base = sst->mailbox.in_base;
 	void __iomem *fw_reg_addr;
@@ -213,7 +213,7 @@ static const struct file_operations soft_regs_ctrl_fops = {
 	.llseek = default_llseek,
 };
 
-struct skl_debug *skl_debugfs_init(struct skl *skl)
+struct skl_debug *skl_debugfs_init(struct skl_dev *skl)
 {
 	struct skl_debug *d;
 
@@ -252,7 +252,7 @@ err:
 	return NULL;
 }
 
-void skl_debugfs_exit(struct skl *skl)
+void skl_debugfs_exit(struct skl_dev *skl)
 {
 	struct skl_debug *d = skl->debugfs;
 
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index febc070839e0..f604d9c6824d 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -47,7 +47,7 @@ static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab)
 
 #define SKL_ASTATE_PARAM_ID	4
 
-void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data)
+void skl_dsp_set_astate_cfg(struct skl_dev *skl, u32 cnt, void *data)
 {
 	struct skl_ipc_large_config_msg	msg = {0};
 
@@ -55,14 +55,14 @@ void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data)
 	msg.param_data_size = (cnt * sizeof(struct skl_astate_param) +
 				sizeof(cnt));
 
-	skl_ipc_set_large_config(&ctx->ipc, &msg, data);
+	skl_ipc_set_large_config(&skl->ipc, &msg, data);
 }
 
 #define NOTIFICATION_PARAM_ID 3
 #define NOTIFICATION_MASK 0xf
 
 /* disable notfication for underruns/overruns from firmware module */
-void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
+void skl_dsp_enable_notification(struct skl_dev *skl, bool enable)
 {
 	struct notification_mask mask;
 	struct skl_ipc_large_config_msg	msg = {0};
@@ -73,7 +73,7 @@ void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
 	msg.large_param_id = NOTIFICATION_PARAM_ID;
 	msg.param_data_size = sizeof(mask);
 
-	skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask);
+	skl_ipc_set_large_config(&skl->ipc, &msg, (u32 *)&mask);
 }
 
 static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
@@ -277,7 +277,7 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id)
 	return NULL;
 }
 
-int skl_init_dsp(struct skl *skl)
+int skl_init_dsp(struct skl_dev *skl)
 {
 	void __iomem *mmio_base;
 	struct hdac_bus *bus = skl_to_bus(skl);
@@ -307,13 +307,13 @@ int skl_init_dsp(struct skl *skl)
 	loader_ops = ops->loader_ops();
 	ret = ops->init(bus->dev, mmio_base, irq,
 				skl->fw_name, loader_ops,
-				&skl->skl_sst);
+				&skl);
 
 	if (ret < 0)
 		goto unmap_mmio;
 
-	skl->skl_sst->dsp_ops = ops;
-	cores = &skl->skl_sst->cores;
+	skl->dsp_ops = ops;
+	cores = &skl->cores;
 	cores->count = ops->num_cores;
 
 	cores->state = kcalloc(cores->count, sizeof(*cores->state), GFP_KERNEL);
@@ -342,21 +342,20 @@ unmap_mmio:
 	return ret;
 }
 
-int skl_free_dsp(struct skl *skl)
+int skl_free_dsp(struct skl_dev *skl)
 {
 	struct hdac_bus *bus = skl_to_bus(skl);
-	struct skl_sst *ctx = skl->skl_sst;
 
 	/* disable  ppcap interrupt */
 	snd_hdac_ext_bus_ppcap_int_enable(bus, false);
 
-	ctx->dsp_ops->cleanup(bus->dev, ctx);
+	skl->dsp_ops->cleanup(bus->dev, skl);
 
-	kfree(ctx->cores.state);
-	kfree(ctx->cores.usage_count);
+	kfree(skl->cores.state);
+	kfree(skl->cores.usage_count);
 
-	if (ctx->dsp->addr.lpe)
-		iounmap(ctx->dsp->addr.lpe);
+	if (skl->dsp->addr.lpe)
+		iounmap(skl->dsp->addr.lpe);
 
 	return 0;
 }
@@ -368,15 +367,14 @@ int skl_free_dsp(struct skl *skl)
  * mode during system suspend. In the case of normal suspend, cancel
  * any pending D0i3 work.
  */
-int skl_suspend_late_dsp(struct skl *skl)
+int skl_suspend_late_dsp(struct skl_dev *skl)
 {
-	struct skl_sst *ctx = skl->skl_sst;
 	struct delayed_work *dwork;
 
-	if (!ctx)
+	if (!skl)
 		return 0;
 
-	dwork = &ctx->d0i3.work;
+	dwork = &skl->d0i3.work;
 
 	if (dwork->work.func) {
 		if (skl->supend_active)
@@ -388,9 +386,8 @@ int skl_suspend_late_dsp(struct skl *skl)
 	return 0;
 }
 
-int skl_suspend_dsp(struct skl *skl)
+int skl_suspend_dsp(struct skl_dev *skl)
 {
-	struct skl_sst *ctx = skl->skl_sst;
 	struct hdac_bus *bus = skl_to_bus(skl);
 	int ret;
 
@@ -398,7 +395,7 @@ int skl_suspend_dsp(struct skl *skl)
 	if (!bus->ppcap)
 		return 0;
 
-	ret = skl_dsp_sleep(ctx->dsp);
+	ret = skl_dsp_sleep(skl->dsp);
 	if (ret < 0)
 		return ret;
 
@@ -409,9 +406,8 @@ int skl_suspend_dsp(struct skl *skl)
 	return 0;
 }
 
-int skl_resume_dsp(struct skl *skl)
+int skl_resume_dsp(struct skl_dev *skl)
 {
-	struct skl_sst *ctx = skl->skl_sst;
 	struct hdac_bus *bus = skl_to_bus(skl);
 	int ret;
 
@@ -424,26 +420,26 @@ int skl_resume_dsp(struct skl *skl)
 	snd_hdac_ext_bus_ppcap_int_enable(bus, true);
 
 	/* check if DSP 1st boot is done */
-	if (skl->skl_sst->is_first_boot)
+	if (skl->is_first_boot)
 		return 0;
 
 	/*
 	 * Disable dynamic clock and power gating during firmware
 	 * and library download
 	 */
-	ctx->enable_miscbdcge(ctx->dev, false);
-	ctx->clock_power_gating(ctx->dev, false);
+	skl->enable_miscbdcge(skl->dev, false);
+	skl->clock_power_gating(skl->dev, false);
 
-	ret = skl_dsp_wake(ctx->dsp);
-	ctx->enable_miscbdcge(ctx->dev, true);
-	ctx->clock_power_gating(ctx->dev, true);
+	ret = skl_dsp_wake(skl->dsp);
+	skl->enable_miscbdcge(skl->dev, true);
+	skl->clock_power_gating(skl->dev, true);
 	if (ret < 0)
 		return ret;
 
-	skl_dsp_enable_notification(skl->skl_sst, false);
+	skl_dsp_enable_notification(skl, false);
 
 	if (skl->cfg.astate_cfg != NULL) {
-		skl_dsp_set_astate_cfg(skl->skl_sst, skl->cfg.astate_cfg->count,
+		skl_dsp_set_astate_cfg(skl, skl->cfg.astate_cfg->count,
 					skl->cfg.astate_cfg);
 	}
 	return ret;
@@ -476,7 +472,7 @@ enum skl_bitdepth skl_get_bit_depth(int params)
  * which are read from widget information passed through topology binary
  * This is send when we create a module with INIT_INSTANCE IPC msg
  */
-static void skl_set_base_module_format(struct skl_sst *ctx,
+static void skl_set_base_module_format(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_base_cfg *base_cfg)
 {
@@ -493,7 +489,7 @@ static void skl_set_base_module_format(struct skl_sst *ctx,
 	base_cfg->audio_fmt.ch_cfg = format->ch_cfg;
 	base_cfg->audio_fmt.sample_type = format->sample_type;
 
-	dev_dbg(ctx->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n",
+	dev_dbg(skl->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n",
 			format->bit_depth, format->valid_bit_depth,
 			format->ch_cfg);
 
@@ -530,7 +526,7 @@ static void skl_copy_copier_caps(struct skl_module_cfg *mconfig,
  * Calculate the gatewat settings required for copier module, type of
  * gateway and index of gateway to use
  */
-static u32 skl_get_node_id(struct skl_sst *ctx,
+static u32 skl_get_node_id(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig)
 {
 	union skl_connector_node_id node_id = {0};
@@ -587,16 +583,15 @@ static u32 skl_get_node_id(struct skl_sst *ctx,
 	return node_id.val;
 }
 
-static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
+static void skl_setup_cpr_gateway_cfg(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_cpr_cfg *cpr_mconfig)
 {
 	u32 dma_io_buf;
 	struct skl_module_res *res;
 	int res_idx = mconfig->res_idx;
-	struct skl *skl = get_skl_ctx(ctx->dev);
 
-	cpr_mconfig->gtw_cfg.node_id = skl_get_node_id(ctx, mconfig);
+	cpr_mconfig->gtw_cfg.node_id = skl_get_node_id(skl, mconfig);
 
 	if (cpr_mconfig->gtw_cfg.node_id == SKL_NON_GATEWAY_CPR_NODE_ID) {
 		cpr_mconfig->cpr_feature_mask = 0;
@@ -627,7 +622,7 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 		break;
 
 	default:
-		dev_warn(ctx->dev, "wrong connection type: %d\n",
+		dev_warn(skl->dev, "wrong connection type: %d\n",
 				mconfig->hw_conn_type);
 		return;
 	}
@@ -653,7 +648,7 @@ skip_buf_size_calc:
 #define DMA_CONTROL_ID 5
 #define DMA_I2S_BLOB_SIZE 21
 
-int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps,
+int skl_dsp_set_dma_control(struct skl_dev *skl, u32 *caps,
 				u32 caps_size, u32 node_id)
 {
 	struct skl_dma_control *dma_ctrl;
@@ -686,14 +681,14 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps,
 
 	memcpy(dma_ctrl->config_data, caps, caps_size);
 
-	err = skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)dma_ctrl);
+	err = skl_ipc_set_large_config(&skl->ipc, &msg, (u32 *)dma_ctrl);
 
 	kfree(dma_ctrl);
 	return err;
 }
 EXPORT_SYMBOL_GPL(skl_dsp_set_dma_control);
 
-static void skl_setup_out_format(struct skl_sst *ctx,
+static void skl_setup_out_format(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_audio_data_format *out_fmt)
 {
@@ -711,7 +706,7 @@ static void skl_setup_out_format(struct skl_sst *ctx,
 	out_fmt->interleaving = format->interleaving_style;
 	out_fmt->sample_type = format->sample_type;
 
-	dev_dbg(ctx->dev, "copier out format chan=%d fre=%d bitdepth=%d\n",
+	dev_dbg(skl->dev, "copier out format chan=%d fre=%d bitdepth=%d\n",
 		out_fmt->number_of_channels, format->s_freq, format->bit_depth);
 }
 
@@ -720,7 +715,7 @@ static void skl_setup_out_format(struct skl_sst *ctx,
  * configuration and the target frequency as extra parameter passed as src
  * config
  */
-static void skl_set_src_format(struct skl_sst *ctx,
+static void skl_set_src_format(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_src_module_cfg *src_mconfig)
 {
@@ -728,7 +723,7 @@ static void skl_set_src_format(struct skl_sst *ctx,
 	struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx];
 	struct skl_module_fmt *fmt = &iface->outputs[0].fmt;
 
-	skl_set_base_module_format(ctx, mconfig,
+	skl_set_base_module_format(skl, mconfig,
 		(struct skl_base_cfg *)src_mconfig);
 
 	src_mconfig->src_cfg = fmt->s_freq;
@@ -739,7 +734,7 @@ static void skl_set_src_format(struct skl_sst *ctx,
  * module configuration and channel configuration
  * It also take coefficients and now we have defaults applied here
  */
-static void skl_set_updown_mixer_format(struct skl_sst *ctx,
+static void skl_set_updown_mixer_format(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_up_down_mixer_cfg *mixer_mconfig)
 {
@@ -747,7 +742,7 @@ static void skl_set_updown_mixer_format(struct skl_sst *ctx,
 	struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx];
 	struct skl_module_fmt *fmt = &iface->outputs[0].fmt;
 
-	skl_set_base_module_format(ctx,	mconfig,
+	skl_set_base_module_format(skl,	mconfig,
 		(struct skl_base_cfg *)mixer_mconfig);
 	mixer_mconfig->out_ch_cfg = fmt->ch_cfg;
 	mixer_mconfig->ch_map = fmt->ch_map;
@@ -760,17 +755,17 @@ static void skl_set_updown_mixer_format(struct skl_sst *ctx,
  * format, gateway settings
  * copier_module_config is sent as input buffer with INIT_INSTANCE IPC msg
  */
-static void skl_set_copier_format(struct skl_sst *ctx,
+static void skl_set_copier_format(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_cpr_cfg *cpr_mconfig)
 {
 	struct skl_audio_data_format *out_fmt = &cpr_mconfig->out_fmt;
 	struct skl_base_cfg *base_cfg = (struct skl_base_cfg *)cpr_mconfig;
 
-	skl_set_base_module_format(ctx, mconfig, base_cfg);
+	skl_set_base_module_format(skl, mconfig, base_cfg);
 
-	skl_setup_out_format(ctx, mconfig, out_fmt);
-	skl_setup_cpr_gateway_cfg(ctx, mconfig, cpr_mconfig);
+	skl_setup_out_format(skl, mconfig, out_fmt);
+	skl_setup_cpr_gateway_cfg(skl, mconfig, cpr_mconfig);
 }
 
 /*
@@ -778,13 +773,13 @@ static void skl_set_copier_format(struct skl_sst *ctx,
  * configuration and params
  */
 
-static void skl_set_algo_format(struct skl_sst *ctx,
+static void skl_set_algo_format(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_algo_cfg *algo_mcfg)
 {
 	struct skl_base_cfg *base_cfg = (struct skl_base_cfg *)algo_mcfg;
 
-	skl_set_base_module_format(ctx, mconfig, base_cfg);
+	skl_set_base_module_format(skl, mconfig, base_cfg);
 
 	if (mconfig->formats_config.caps_size == 0)
 		return;
@@ -802,7 +797,7 @@ static void skl_set_algo_format(struct skl_sst *ctx,
  * Mic select module take base module configuration and out-format
  * configuration
  */
-static void skl_set_base_outfmt_format(struct skl_sst *ctx,
+static void skl_set_base_outfmt_format(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig,
 			struct skl_base_outfmt_cfg *base_outfmt_mcfg)
 {
@@ -810,11 +805,11 @@ static void skl_set_base_outfmt_format(struct skl_sst *ctx,
 	struct skl_base_cfg *base_cfg =
 				(struct skl_base_cfg *)base_outfmt_mcfg;
 
-	skl_set_base_module_format(ctx, mconfig, base_cfg);
-	skl_setup_out_format(ctx, mconfig, out_fmt);
+	skl_set_base_module_format(skl, mconfig, base_cfg);
+	skl_setup_out_format(skl, mconfig, out_fmt);
 }
 
-static u16 skl_get_module_param_size(struct skl_sst *ctx,
+static u16 skl_get_module_param_size(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig)
 {
 	u16 param_size;
@@ -859,14 +854,14 @@ static u16 skl_get_module_param_size(struct skl_sst *ctx,
  * base module format configuration
  */
 
-static int skl_set_module_format(struct skl_sst *ctx,
+static int skl_set_module_format(struct skl_dev *skl,
 			struct skl_module_cfg *module_config,
 			u16 *module_config_size,
 			void **param_data)
 {
 	u16 param_size;
 
-	param_size  = skl_get_module_param_size(ctx, module_config);
+	param_size  = skl_get_module_param_size(skl, module_config);
 
 	*param_data = kzalloc(param_size, GFP_KERNEL);
 	if (NULL == *param_data)
@@ -876,34 +871,34 @@ static int skl_set_module_format(struct skl_sst *ctx,
 
 	switch (module_config->m_type) {
 	case SKL_MODULE_TYPE_COPIER:
-		skl_set_copier_format(ctx, module_config, *param_data);
+		skl_set_copier_format(skl, module_config, *param_data);
 		break;
 
 	case SKL_MODULE_TYPE_SRCINT:
-		skl_set_src_format(ctx, module_config, *param_data);
+		skl_set_src_format(skl, module_config, *param_data);
 		break;
 
 	case SKL_MODULE_TYPE_UPDWMIX:
-		skl_set_updown_mixer_format(ctx, module_config, *param_data);
+		skl_set_updown_mixer_format(skl, module_config, *param_data);
 		break;
 
 	case SKL_MODULE_TYPE_ALGO:
-		skl_set_algo_format(ctx, module_config, *param_data);
+		skl_set_algo_format(skl, module_config, *param_data);
 		break;
 
 	case SKL_MODULE_TYPE_BASE_OUTFMT:
 	case SKL_MODULE_TYPE_MIC_SELECT:
 	case SKL_MODULE_TYPE_KPB:
-		skl_set_base_outfmt_format(ctx, module_config, *param_data);
+		skl_set_base_outfmt_format(skl, module_config, *param_data);
 		break;
 
 	default:
-		skl_set_base_module_format(ctx, module_config, *param_data);
+		skl_set_base_module_format(skl, module_config, *param_data);
 		break;
 
 	}
 
-	dev_dbg(ctx->dev, "Module type=%d config size: %d bytes\n",
+	dev_dbg(skl->dev, "Module type=%d config size: %d bytes\n",
 			module_config->id.module_id, param_size);
 	print_hex_dump_debug("Module params:", DUMP_PREFIX_OFFSET, 8, 4,
 			*param_data, param_size, false);
@@ -1004,7 +999,7 @@ static void skl_clear_module_state(struct skl_module_pin *mpin, int max,
  * We first calculate the module format, based on module type and then
  * invoke the DSP by sending IPC INIT_INSTANCE using ipc helper
  */
-int skl_init_module(struct skl_sst *ctx,
+int skl_init_module(struct skl_dev *skl,
 			struct skl_module_cfg *mconfig)
 {
 	u16 module_config_size = 0;
@@ -1012,19 +1007,19 @@ int skl_init_module(struct skl_sst *ctx,
 	int ret;
 	struct skl_ipc_init_instance_msg msg;
 
-	dev_dbg(ctx->dev, "%s: module_id = %d instance=%d\n", __func__,
+	dev_dbg(skl->dev, "%s: module_id = %d instance=%d\n", __func__,
 		 mconfig->id.module_id, mconfig->id.pvt_id);
 
 	if (mconfig->pipe->state != SKL_PIPE_CREATED) {
-		dev_err(ctx->dev, "Pipe not created state= %d pipe_id= %d\n",
+		dev_err(skl->dev, "Pipe not created state= %d pipe_id= %d\n",
 				 mconfig->pipe->state, mconfig->pipe->ppl_id);
 		return -EIO;
 	}
 
-	ret = skl_set_module_format(ctx, mconfig,
+	ret = skl_set_module_format(skl, mconfig,
 			&module_config_size, &param_data);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Failed to set module format ret=%d\n", ret);
+		dev_err(skl->dev, "Failed to set module format ret=%d\n", ret);
 		return ret;
 	}
 
@@ -1035,9 +1030,9 @@ int skl_init_module(struct skl_sst *ctx,
 	msg.core_id = mconfig->core_id;
 	msg.domain = mconfig->domain;
 
-	ret = skl_ipc_init_instance(&ctx->ipc, &msg, param_data);
+	ret = skl_ipc_init_instance(&skl->ipc, &msg, param_data);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Failed to init instance ret=%d\n", ret);
+		dev_err(skl->dev, "Failed to init instance ret=%d\n", ret);
 		kfree(param_data);
 		return ret;
 	}
@@ -1046,15 +1041,15 @@ int skl_init_module(struct skl_sst *ctx,
 	return ret;
 }
 
-static void skl_dump_bind_info(struct skl_sst *ctx, struct skl_module_cfg
+static void skl_dump_bind_info(struct skl_dev *skl, struct skl_module_cfg
 	*src_module, struct skl_module_cfg *dst_module)
 {
-	dev_dbg(ctx->dev, "%s: src module_id = %d  src_instance=%d\n",
+	dev_dbg(skl->dev, "%s: src module_id = %d  src_instance=%d\n",
 		__func__, src_module->id.module_id, src_module->id.pvt_id);
-	dev_dbg(ctx->dev, "%s: dst_module=%d dst_instance=%d\n", __func__,
+	dev_dbg(skl->dev, "%s: dst_module=%d dst_instance=%d\n", __func__,
 		 dst_module->id.module_id, dst_module->id.pvt_id);
 
-	dev_dbg(ctx->dev, "src_module state = %d dst module state = %d\n",
+	dev_dbg(skl->dev, "src_module state = %d dst module state = %d\n",
 		src_module->m_state, dst_module->m_state);
 }
 
@@ -1063,7 +1058,7 @@ static void skl_dump_bind_info(struct skl_sst *ctx, struct skl_module_cfg
  * it is already bind.
  * Find the pin allocated and unbind then using bind_unbind IPC
  */
-int skl_unbind_modules(struct skl_sst *ctx,
+int skl_unbind_modules(struct skl_dev *skl,
 			struct skl_module_cfg *src_mcfg,
 			struct skl_module_cfg *dst_mcfg)
 {
@@ -1075,7 +1070,7 @@ int skl_unbind_modules(struct skl_sst *ctx,
 	int out_max = src_mcfg->module->max_output_pins;
 	int src_index, dst_index, src_pin_state, dst_pin_state;
 
-	skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
+	skl_dump_bind_info(skl, src_mcfg, dst_mcfg);
 
 	/* get src queue index */
 	src_index = skl_get_queue_index(src_mcfg->m_out_pin, dst_id, out_max);
@@ -1104,7 +1099,7 @@ int skl_unbind_modules(struct skl_sst *ctx,
 	msg.dst_instance_id = dst_mcfg->id.pvt_id;
 	msg.bind = false;
 
-	ret = skl_ipc_bind_unbind(&ctx->ipc, &msg);
+	ret = skl_ipc_bind_unbind(&skl->ipc, &msg);
 	if (!ret) {
 		/* free queue only if unbind is success */
 		skl_free_queue(src_mcfg->m_out_pin, src_index);
@@ -1142,7 +1137,7 @@ static void fill_pin_params(struct skl_audio_data_format *pin_fmt,
  * This function finds the pins and then sends bund_unbind IPC message to
  * DSP using IPC helper
  */
-int skl_bind_modules(struct skl_sst *ctx,
+int skl_bind_modules(struct skl_dev *skl,
 			struct skl_module_cfg *src_mcfg,
 			struct skl_module_cfg *dst_mcfg)
 {
@@ -1156,7 +1151,7 @@ int skl_bind_modules(struct skl_sst *ctx,
 	struct skl_module *module;
 	struct skl_module_iface *fmt;
 
-	skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
+	skl_dump_bind_info(skl, src_mcfg, dst_mcfg);
 
 	if (src_mcfg->m_state < SKL_MODULE_INIT_DONE ||
 		dst_mcfg->m_state < SKL_MODULE_INIT_DONE)
@@ -1188,7 +1183,7 @@ int skl_bind_modules(struct skl_sst *ctx,
 
 		format = &fmt->outputs[src_index].fmt;
 		fill_pin_params(&(pin_fmt.dst_fmt), format);
-		ret = skl_set_module_params(ctx, (void *)&pin_fmt,
+		ret = skl_set_module_params(skl, (void *)&pin_fmt,
 					sizeof(struct skl_cpr_pin_fmt),
 					CPR_SINK_FMT_PARAM_ID, src_mcfg);
 
@@ -1198,7 +1193,7 @@ int skl_bind_modules(struct skl_sst *ctx,
 
 	msg.dst_queue = dst_index;
 
-	dev_dbg(ctx->dev, "src queue = %d dst queue =%d\n",
+	dev_dbg(skl->dev, "src queue = %d dst queue =%d\n",
 			 msg.src_queue, msg.dst_queue);
 
 	msg.module_id = src_mcfg->id.module_id;
@@ -1207,7 +1202,7 @@ int skl_bind_modules(struct skl_sst *ctx,
 	msg.dst_instance_id = dst_mcfg->id.pvt_id;
 	msg.bind = true;
 
-	ret = skl_ipc_bind_unbind(&ctx->ipc, &msg);
+	ret = skl_ipc_bind_unbind(&skl->ipc, &msg);
 
 	if (!ret) {
 		src_mcfg->m_state = SKL_MODULE_BIND_DONE;
@@ -1223,12 +1218,12 @@ out:
 	return ret;
 }
 
-static int skl_set_pipe_state(struct skl_sst *ctx, struct skl_pipe *pipe,
+static int skl_set_pipe_state(struct skl_dev *skl, struct skl_pipe *pipe,
 	enum skl_ipc_pipeline_state state)
 {
-	dev_dbg(ctx->dev, "%s: pipe_state = %d\n", __func__, state);
+	dev_dbg(skl->dev, "%s: pipe_state = %d\n", __func__, state);
 
-	return skl_ipc_set_pipeline_state(&ctx->ipc, pipe->ppl_id, state);
+	return skl_ipc_set_pipeline_state(&skl->ipc, pipe->ppl_id, state);
 }
 
 /*
@@ -1237,17 +1232,17 @@ static int skl_set_pipe_state(struct skl_sst *ctx, struct skl_pipe *pipe,
  * This function creates pipeline, by sending create pipeline IPC messages
  * to FW
  */
-int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe)
+int skl_create_pipeline(struct skl_dev *skl, struct skl_pipe *pipe)
 {
 	int ret;
 
-	dev_dbg(ctx->dev, "%s: pipe_id = %d\n", __func__, pipe->ppl_id);
+	dev_dbg(skl->dev, "%s: pipe_id = %d\n", __func__, pipe->ppl_id);
 
-	ret = skl_ipc_create_pipeline(&ctx->ipc, pipe->memory_pages,
+	ret = skl_ipc_create_pipeline(&skl->ipc, pipe->memory_pages,
 				pipe->pipe_priority, pipe->ppl_id,
 				pipe->lp_mode);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Failed to create pipeline\n");
+		dev_err(skl->dev, "Failed to create pipeline\n");
 		return ret;
 	}
 
@@ -1262,11 +1257,11 @@ int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe)
  * reset state. Finish the procedure by sending delete pipeline IPC.
  * DSP will stop the DMA engines and release resources
  */
-int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
+int skl_delete_pipe(struct skl_dev *skl, struct skl_pipe *pipe)
 {
 	int ret;
 
-	dev_dbg(ctx->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id);
+	dev_dbg(skl->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id);
 
 	/* If pipe was not created in FW, do not try to delete it */
 	if (pipe->state < SKL_PIPE_CREATED)
@@ -1274,9 +1269,9 @@ int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
 
 	/* If pipe is started, do stop the pipe in FW. */
 	if (pipe->state >= SKL_PIPE_STARTED) {
-		ret = skl_set_pipe_state(ctx, pipe, PPL_PAUSED);
+		ret = skl_set_pipe_state(skl, pipe, PPL_PAUSED);
 		if (ret < 0) {
-			dev_err(ctx->dev, "Failed to stop pipeline\n");
+			dev_err(skl->dev, "Failed to stop pipeline\n");
 			return ret;
 		}
 
@@ -1284,17 +1279,17 @@ int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
 	}
 
 	/* reset pipe state before deletion */
-	ret = skl_set_pipe_state(ctx, pipe, PPL_RESET);
+	ret = skl_set_pipe_state(skl, pipe, PPL_RESET);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Failed to reset pipe ret=%d\n", ret);
+		dev_err(skl->dev, "Failed to reset pipe ret=%d\n", ret);
 		return ret;
 	}
 
 	pipe->state = SKL_PIPE_RESET;
 
-	ret = skl_ipc_delete_pipeline(&ctx->ipc, pipe->ppl_id);
+	ret = skl_ipc_delete_pipeline(&skl->ipc, pipe->ppl_id);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Failed to delete pipeline\n");
+		dev_err(skl->dev, "Failed to delete pipeline\n");
 		return ret;
 	}
 
@@ -1308,28 +1303,28 @@ int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
  * For processing data the pipe need to be run by sending IPC set pipe state
  * to DSP
  */
-int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
+int skl_run_pipe(struct skl_dev *skl, struct skl_pipe *pipe)
 {
 	int ret;
 
-	dev_dbg(ctx->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id);
+	dev_dbg(skl->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id);
 
 	/* If pipe was not created in FW, do not try to pause or delete */
 	if (pipe->state < SKL_PIPE_CREATED)
 		return 0;
 
 	/* Pipe has to be paused before it is started */
-	ret = skl_set_pipe_state(ctx, pipe, PPL_PAUSED);
+	ret = skl_set_pipe_state(skl, pipe, PPL_PAUSED);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Failed to pause pipe\n");
+		dev_err(skl->dev, "Failed to pause pipe\n");
 		return ret;
 	}
 
 	pipe->state = SKL_PIPE_PAUSED;
 
-	ret = skl_set_pipe_state(ctx, pipe, PPL_RUNNING);
+	ret = skl_set_pipe_state(skl, pipe, PPL_RUNNING);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Failed to start pipe\n");
+		dev_err(skl->dev, "Failed to start pipe\n");
 		return ret;
 	}
 
@@ -1342,19 +1337,19 @@ int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
  * Stop the pipeline by sending set pipe state IPC
  * DSP doesnt implement stop so we always send pause message
  */
-int skl_stop_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
+int skl_stop_pipe(struct skl_dev *skl, struct skl_pipe *pipe)
 {
 	int ret;
 
-	dev_dbg(ctx->dev, "In %s pipe=%d\n", __func__, pipe->ppl_id);
+	dev_dbg(skl->dev, "In %s pipe=%d\n", __func__, pipe->ppl_id);
 
 	/* If pipe was not created in FW, do not try to pause or delete */
 	if (pipe->state < SKL_PIPE_PAUSED)
 		return 0;
 
-	ret = skl_set_pipe_state(ctx, pipe, PPL_PAUSED);
+	ret = skl_set_pipe_state(skl, pipe, PPL_PAUSED);
 	if (ret < 0) {
-		dev_dbg(ctx->dev, "Failed to stop pipe\n");
+		dev_dbg(skl->dev, "Failed to stop pipe\n");
 		return ret;
 	}
 
@@ -1367,7 +1362,7 @@ int skl_stop_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
  * Reset the pipeline by sending set pipe state IPC this will reset the DMA
  * from the DSP side
  */
-int skl_reset_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
+int skl_reset_pipe(struct skl_dev *skl, struct skl_pipe *pipe)
 {
 	int ret;
 
@@ -1375,9 +1370,9 @@ int skl_reset_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
 	if (pipe->state < SKL_PIPE_PAUSED)
 		return 0;
 
-	ret = skl_set_pipe_state(ctx, pipe, PPL_RESET);
+	ret = skl_set_pipe_state(skl, pipe, PPL_RESET);
 	if (ret < 0) {
-		dev_dbg(ctx->dev, "Failed to reset pipe ret=%d\n", ret);
+		dev_dbg(skl->dev, "Failed to reset pipe ret=%d\n", ret);
 		return ret;
 	}
 
@@ -1387,7 +1382,7 @@ int skl_reset_pipe(struct skl_sst *ctx, struct skl_pipe *pipe)
 }
 
 /* Algo parameter set helper function */
-int skl_set_module_params(struct skl_sst *ctx, u32 *params, int size,
+int skl_set_module_params(struct skl_dev *skl, u32 *params, int size,
 				u32 param_id, struct skl_module_cfg *mcfg)
 {
 	struct skl_ipc_large_config_msg msg;
@@ -1397,10 +1392,10 @@ int skl_set_module_params(struct skl_sst *ctx, u32 *params, int size,
 	msg.param_data_size = size;
 	msg.large_param_id = param_id;
 
-	return skl_ipc_set_large_config(&ctx->ipc, &msg, params);
+	return skl_ipc_set_large_config(&skl->ipc, &msg, params);
 }
 
-int skl_get_module_params(struct skl_sst *ctx, u32 *params, int size,
+int skl_get_module_params(struct skl_dev *skl, u32 *params, int size,
 			  u32 param_id, struct skl_module_cfg *mcfg)
 {
 	struct skl_ipc_large_config_msg msg;
@@ -1410,5 +1405,5 @@ int skl_get_module_params(struct skl_sst *ctx, u32 *params, int size,
 	msg.param_data_size = size;
 	msg.large_param_id = param_id;
 
-	return skl_ipc_get_large_config(&ctx->ipc, &msg, params);
+	return skl_ipc_get_large_config(&skl->ipc, &msg, params);
 }
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 1132109cb992..6fc3a190067e 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -126,7 +126,7 @@ static bool skl_check_ep_match(struct device *dev, struct nhlt_endpoint *epnt,
 }
 
 struct nhlt_specific_cfg
-*skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type,
+*skl_get_ep_blob(struct skl_dev *skl, u32 instance, u8 link_type,
 			u8 s_fmt, u8 num_ch, u32 s_rate,
 			u8 dirn, u8 dev_type)
 {
@@ -162,7 +162,7 @@ struct nhlt_specific_cfg
 	return NULL;
 }
 
-int skl_get_dmic_geo(struct skl *skl)
+int skl_get_dmic_geo(struct skl_dev *skl)
 {
 	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
 	struct nhlt_endpoint *epnt;
@@ -219,7 +219,7 @@ static void skl_nhlt_trim_space(char *trim)
 	s[cnt] = '\0';
 }
 
-int skl_nhlt_update_topology_bin(struct skl *skl)
+int skl_nhlt_update_topology_bin(struct skl_dev *skl)
 {
 	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
 	struct hdac_bus *bus = skl_to_bus(skl);
@@ -243,7 +243,7 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev,
 {
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct hdac_bus *bus = pci_get_drvdata(pci);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
 	char platform_id[32];
 
@@ -257,7 +257,7 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev,
 
 static DEVICE_ATTR(platform_id, 0444, skl_nhlt_platform_id_show, NULL);
 
-int skl_nhlt_create_sysfs(struct skl *skl)
+int skl_nhlt_create_sysfs(struct skl_dev *skl)
 {
 	struct device *dev = &skl->pci->dev;
 
@@ -267,7 +267,7 @@ int skl_nhlt_create_sysfs(struct skl *skl)
 	return 0;
 }
 
-void skl_nhlt_remove_sysfs(struct skl *skl)
+void skl_nhlt_remove_sysfs(struct skl_dev *skl)
 {
 	struct device *dev = &skl->pci->dev;
 
@@ -279,7 +279,7 @@ void skl_nhlt_remove_sysfs(struct skl *skl)
  * stores all possible rates supported in a rate table for the corresponding
  * sclk/sclkfs.
  */
-static void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks,
+static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 				struct nhlt_fmt *fmt, u8 id)
 {
 	struct skl_i2s_config_blob_ext *i2s_config_ext;
@@ -377,7 +377,7 @@ static void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks,
 	}
 }
 
-static void skl_get_mclk(struct skl *skl, struct skl_ssp_clk *mclk,
+static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
 				struct nhlt_fmt *fmt, u8 id)
 {
 	struct skl_i2s_config_blob_ext *i2s_config_ext;
@@ -421,7 +421,7 @@ static void skl_get_mclk(struct skl *skl, struct skl_ssp_clk *mclk,
 	mclk[id].parent_name = parent->name;
 }
 
-void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks)
+void skl_get_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks)
 {
 	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
 	struct nhlt_endpoint *epnt;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 760bbcf9a469..2ae5d09d7228 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -116,7 +116,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream,
 {
 	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct snd_soc_dapm_widget *w;
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		w = dai->playback_widget;
@@ -132,7 +132,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream,
 int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params)
 {
 	struct hdac_bus *bus = dev_get_drvdata(dev);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	unsigned int format_val;
 	struct hdac_stream *hstream;
 	struct hdac_ext_stream *stream;
@@ -224,7 +224,7 @@ static int skl_pcm_open(struct snd_pcm_substream *substream,
 	struct hdac_ext_stream *stream;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct skl_dma_params *dma_params;
-	struct skl *skl = get_skl_ctx(dai->dev);
+	struct skl_dev *skl = get_skl_ctx(dai->dev);
 	struct skl_module_cfg *mconfig;
 
 	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
@@ -271,7 +271,7 @@ static int skl_pcm_open(struct snd_pcm_substream *substream,
 static int skl_pcm_prepare(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
-	struct skl *skl = get_skl_ctx(dai->dev);
+	struct skl_dev *skl = get_skl_ctx(dai->dev);
 	struct skl_module_cfg *mconfig;
 	int ret;
 
@@ -288,7 +288,7 @@ static int skl_pcm_prepare(struct snd_pcm_substream *substream,
 		 mconfig->pipe->state == SKL_PIPE_CREATED ||
 		 mconfig->pipe->state == SKL_PIPE_PAUSED)) {
 
-		ret = skl_reset_pipe(skl->skl_sst, mconfig->pipe);
+		ret = skl_reset_pipe(skl, mconfig->pipe);
 
 		if (ret < 0)
 			return ret;
@@ -350,7 +350,7 @@ static void skl_pcm_close(struct snd_pcm_substream *substream,
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
 	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct skl_dma_params *dma_params = NULL;
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct skl_module_cfg *mconfig;
 
 	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
@@ -370,9 +370,9 @@ static void skl_pcm_close(struct snd_pcm_substream *substream,
 	 * CGCTL.MISCBDCGE if disabled by driver
 	 */
 	if (!strncmp(dai->name, "Reference Pin", 13) &&
-			skl->skl_sst->miscbdcg_disabled) {
-		skl->skl_sst->enable_miscbdcge(dai->dev, true);
-		skl->skl_sst->miscbdcg_disabled = false;
+			skl->miscbdcg_disabled) {
+		skl->enable_miscbdcge(dai->dev, true);
+		skl->miscbdcg_disabled = false;
 	}
 
 	mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
@@ -387,7 +387,7 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream,
 {
 	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
-	struct skl *skl = get_skl_ctx(dai->dev);
+	struct skl_dev *skl = get_skl_ctx(dai->dev);
 	struct skl_module_cfg *mconfig;
 	int ret;
 
@@ -396,7 +396,7 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream,
 	mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
 
 	if (mconfig) {
-		ret = skl_reset_pipe(skl->skl_sst, mconfig->pipe);
+		ret = skl_reset_pipe(skl, mconfig->pipe);
 		if (ret < 0)
 			dev_err(dai->dev, "%s:Reset failed ret =%d",
 						__func__, ret);
@@ -471,8 +471,7 @@ static int skl_decoupled_trigger(struct snd_pcm_substream *substream,
 static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 		struct snd_soc_dai *dai)
 {
-	struct skl *skl = get_skl_ctx(dai->dev);
-	struct skl_sst *ctx = skl->skl_sst;
+	struct skl_dev *skl = get_skl_ctx(dai->dev);
 	struct skl_module_cfg *mconfig;
 	struct hdac_bus *bus = get_bus_ctx(substream);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
@@ -515,7 +514,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 		ret = skl_decoupled_trigger(substream, cmd);
 		if (ret < 0)
 			return ret;
-		return skl_run_pipe(ctx, mconfig->pipe);
+		return skl_run_pipe(skl, mconfig->pipe);
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -526,7 +525,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 		 * there are no underrun/overrun in the case if there is a delay
 		 * between the two operations.
 		 */
-		ret = skl_stop_pipe(ctx, mconfig->pipe);
+		ret = skl_stop_pipe(skl, mconfig->pipe);
 		if (ret < 0)
 			return ret;
 
@@ -602,14 +601,14 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
 static int skl_link_pcm_prepare(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
-	struct skl *skl = get_skl_ctx(dai->dev);
+	struct skl_dev *skl = get_skl_ctx(dai->dev);
 	struct skl_module_cfg *mconfig = NULL;
 
 	/* In case of XRUN recovery, reset the FW pipe to clean state */
 	mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream);
 	if (mconfig && !mconfig->pipe->passthru &&
 		(substream->runtime->status->state == SNDRV_PCM_STATE_XRUN))
-		skl_reset_pipe(skl->skl_sst, mconfig->pipe);
+		skl_reset_pipe(skl, mconfig->pipe);
 
 	return 0;
 }
@@ -1301,7 +1300,7 @@ static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct snd_pcm *pcm = rtd->pcm;
 	unsigned int size;
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	if (dai->driver->playback.channels_min ||
 		dai->driver->capture.channels_min) {
@@ -1318,9 +1317,9 @@ static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig)
+static int skl_get_module_info(struct skl_dev *skl,
+		struct skl_module_cfg *mconfig)
 {
-	struct skl_sst *ctx = skl->skl_sst;
 	struct skl_module_inst_id *pin_id;
 	guid_t *uuid_mod, *uuid_tplg;
 	struct skl_module *skl_module;
@@ -1329,12 +1328,12 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig)
 
 	uuid_mod = (guid_t *)mconfig->guid;
 
-	if (list_empty(&ctx->uuid_list)) {
-		dev_err(ctx->dev, "Module list is empty\n");
+	if (list_empty(&skl->uuid_list)) {
+		dev_err(skl->dev, "Module list is empty\n");
 		return -EIO;
 	}
 
-	list_for_each_entry(module, &ctx->uuid_list, list) {
+	list_for_each_entry(module, &skl->uuid_list, list) {
 		if (guid_equal(uuid_mod, &module->uuid)) {
 			mconfig->id.module_id = module->id;
 			if (mconfig->module)
@@ -1361,7 +1360,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig)
 	if (skl->nr_modules && ret)
 		return ret;
 
-	list_for_each_entry(module, &ctx->uuid_list, list) {
+	list_for_each_entry(module, &skl->uuid_list, list) {
 		for (i = 0; i < MAX_IN_QUEUE; i++) {
 			pin_id = &mconfig->m_in_pin[i].id;
 			if (guid_equal(&pin_id->mod_uuid, &module->uuid))
@@ -1378,7 +1377,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig)
 	return 0;
 }
 
-static int skl_populate_modules(struct skl *skl)
+static int skl_populate_modules(struct skl_dev *skl)
 {
 	struct skl_pipeline *p;
 	struct skl_pipe_module *m;
@@ -1393,7 +1392,7 @@ static int skl_populate_modules(struct skl *skl)
 
 			ret = skl_get_module_info(skl, mconfig);
 			if (ret < 0) {
-				dev_err(skl->skl_sst->dev,
+				dev_err(skl->dev,
 					"query module info failed\n");
 				return ret;
 			}
@@ -1408,7 +1407,7 @@ static int skl_populate_modules(struct skl *skl)
 static int skl_platform_soc_probe(struct snd_soc_component *component)
 {
 	struct hdac_bus *bus = dev_get_drvdata(component->dev);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	const struct skl_dsp_ops *ops;
 	int ret;
 
@@ -1434,22 +1433,22 @@ static int skl_platform_soc_probe(struct snd_soc_component *component)
 		 * Disable dynamic clock and power gating during firmware
 		 * and library download
 		 */
-		skl->skl_sst->enable_miscbdcge(component->dev, false);
-		skl->skl_sst->clock_power_gating(component->dev, false);
+		skl->enable_miscbdcge(component->dev, false);
+		skl->clock_power_gating(component->dev, false);
 
-		ret = ops->init_fw(component->dev, skl->skl_sst);
-		skl->skl_sst->enable_miscbdcge(component->dev, true);
-		skl->skl_sst->clock_power_gating(component->dev, true);
+		ret = ops->init_fw(component->dev, skl);
+		skl->enable_miscbdcge(component->dev, true);
+		skl->clock_power_gating(component->dev, true);
 		if (ret < 0) {
 			dev_err(component->dev, "Failed to boot first fw: %d\n", ret);
 			return ret;
 		}
 		skl_populate_modules(skl);
-		skl->skl_sst->update_d0i3c = skl_update_d0i3c;
-		skl_dsp_enable_notification(skl->skl_sst, false);
+		skl->update_d0i3c = skl_update_d0i3c;
+		skl_dsp_enable_notification(skl, false);
 
 		if (skl->cfg.astate_cfg != NULL) {
-			skl_dsp_set_astate_cfg(skl->skl_sst,
+			skl_dsp_set_astate_cfg(skl,
 					skl->cfg.astate_cfg->count,
 					skl->cfg.astate_cfg);
 		}
@@ -1463,7 +1462,7 @@ static int skl_platform_soc_probe(struct snd_soc_component *component)
 static void skl_pcm_remove(struct snd_soc_component *component)
 {
 	struct hdac_bus *bus = dev_get_drvdata(component->dev);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	skl_tplg_exit(component, bus);
 
@@ -1486,7 +1485,7 @@ int skl_platform_register(struct device *dev)
 	struct snd_soc_dai_driver *dais;
 	int num_dais = ARRAY_SIZE(skl_platform_dai);
 	struct hdac_bus *bus = dev_get_drvdata(dev);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	skl->dais = kmemdup(skl_platform_dai, sizeof(skl_platform_dai),
 			    GFP_KERNEL);
@@ -1520,7 +1519,7 @@ err:
 int skl_platform_unregister(struct device *dev)
 {
 	struct hdac_bus *bus = dev_get_drvdata(dev);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct skl_module_deferred_bind *modules, *tmp;
 
 	if (!list_empty(&skl->bind_list)) {
diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c
index 5bb6e40d4d3e..1608d21bcba1 100644
--- a/sound/soc/intel/skylake/skl-ssp-clk.c
+++ b/sound/soc/intel/skylake/skl-ssp-clk.c
@@ -101,7 +101,7 @@ static void skl_fill_clk_ipc(struct skl_clk_rate_cfg_table *rcfg, u8 clk_type)
 }
 
 /* Sends dma control IPC to turn the clock ON/OFF */
-static int skl_send_clk_dma_control(struct skl *skl,
+static int skl_send_clk_dma_control(struct skl_dev *skl,
 				struct skl_clk_rate_cfg_table *rcfg,
 				u32 vbus_id, u8 clk_type,
 				bool enable)
@@ -152,7 +152,7 @@ static int skl_send_clk_dma_control(struct skl *skl,
 	memcpy(i2s_config + sp_cfg->size, data, size);
 
 	node_id = ((SKL_DMA_I2S_LINK_INPUT_CLASS << 8) | (vbus_id << 4));
-	ret = skl_dsp_set_dma_control(skl->skl_sst, (u32 *)i2s_config,
+	ret = skl_dsp_set_dma_control(skl, (u32 *)i2s_config,
 					i2s_config_size, node_id);
 	kfree(i2s_config);
 
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index 36590c5b4673..225706d148d8 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -12,7 +12,7 @@
 #include "../common/sst-dsp.h"
 #include "../common/sst-ipc.h"
 #include "../common/sst-dsp-priv.h"
-#include "skl-sst-ipc.h"
+#include "skl.h"
 
 /* various timeout values */
 #define SKL_DSP_PU_TO		50
@@ -33,7 +33,7 @@ void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state)
  */
 void skl_dsp_init_core_state(struct sst_dsp *ctx)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	int i;
 
 	skl->cores.state[SKL_DSP_CORE0_ID] = SKL_DSP_RUNNING;
@@ -48,7 +48,7 @@ void skl_dsp_init_core_state(struct sst_dsp *ctx)
 /* Get the mask for all enabled cores */
 unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	unsigned int core_mask, en_cores_mask;
 	u32 val;
 
@@ -335,7 +335,7 @@ irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id)
  */
 int skl_dsp_get_core(struct sst_dsp *ctx, unsigned int core_id)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	int ret = 0;
 
 	if (core_id >= skl->cores.count) {
@@ -364,7 +364,7 @@ EXPORT_SYMBOL_GPL(skl_dsp_get_core);
 
 int skl_dsp_put_core(struct sst_dsp *ctx, unsigned int core_id)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	int ret = 0;
 
 	if (core_id >= skl->cores.count) {
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index a80219562036..ed968f1042c7 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -15,9 +15,9 @@
 #include "skl-sst-cldma.h"
 
 struct sst_dsp;
-struct skl_sst;
 struct sst_dsp_device;
 struct skl_lib_info;
+struct skl_dev;
 
 /* Intel HD Audio General DSP Registers */
 #define SKL_ADSP_GEN_BASE		0x0
@@ -222,32 +222,32 @@ int skl_dsp_put_core(struct sst_dsp *ctx, unsigned int core_id);
 int skl_dsp_boot(struct sst_dsp *ctx);
 int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 		const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
-		struct skl_sst **dsp);
+		struct skl_dev **dsp);
 int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 		const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
-		struct skl_sst **dsp);
-int skl_sst_init_fw(struct device *dev, struct skl_sst *ctx);
-int bxt_sst_init_fw(struct device *dev, struct skl_sst *ctx);
-void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
-void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
+		struct skl_dev **dsp);
+int skl_sst_init_fw(struct device *dev, struct skl_dev *skl);
+int bxt_sst_init_fw(struct device *dev, struct skl_dev *skl);
+void skl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl);
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl);
 
 int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw,
 				unsigned int offset, int index);
-int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id);
-int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id);
-int skl_get_pvt_instance_id_map(struct skl_sst *ctx,
+int skl_get_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int instance_id);
+int skl_put_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int *pvt_id);
+int skl_get_pvt_instance_id_map(struct skl_dev *skl,
 				int module_id, int instance_id);
-void skl_freeup_uuid_list(struct skl_sst *ctx);
+void skl_freeup_uuid_list(struct skl_dev *skl);
 
 int skl_dsp_strip_extended_manifest(struct firmware *fw);
-void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable);
+void skl_dsp_enable_notification(struct skl_dev *skl, bool enable);
 
-void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data);
+void skl_dsp_set_astate_cfg(struct skl_dev *skl, u32 cnt, void *data);
 
 int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name,
-		struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp,
+		struct skl_dsp_loader_ops dsp_ops, struct skl_dev **dsp,
 		struct sst_dsp_device *skl_dev);
-int skl_prepare_lib_load(struct skl_sst *skl, struct skl_lib_info *linfo,
+int skl_prepare_lib_load(struct skl_dev *skl, struct skl_lib_info *linfo,
 			struct firmware *stripped_fw,
 			unsigned int hdr_offset, int index);
 void skl_release_library(struct skl_lib_info *linfo, int lib_count);
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index 2cc8f7d2d319..ee1493acc9a8 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -345,7 +345,7 @@ out:
 int skl_ipc_process_notification(struct sst_generic_ipc *ipc,
 		struct skl_ipc_header header)
 {
-	struct skl_sst *skl = container_of(ipc, struct skl_sst, ipc);
+	struct skl_dev *skl = container_of(ipc, struct skl_dev, ipc);
 
 	if (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) {
 		switch (IPC_GLB_NOTIFY_TYPE(header.primary)) {
@@ -436,7 +436,7 @@ void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
 	struct ipc_message *msg;
 	u32 reply = header.primary & IPC_GLB_REPLY_STATUS_MASK;
 	u64 *ipc_header = (u64 *)(&header);
-	struct skl_sst *skl = container_of(ipc, struct skl_sst, ipc);
+	struct skl_dev *skl = container_of(ipc, struct skl_dev, ipc);
 	unsigned long flags;
 
 	spin_lock_irqsave(&ipc->dsp->spinlock, flags);
@@ -488,7 +488,7 @@ void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
 irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context)
 {
 	struct sst_dsp *dsp = context;
-	struct skl_sst *skl = sst_dsp_get_thread_context(dsp);
+	struct skl_dev *skl = sst_dsp_get_thread_context(dsp);
 	struct sst_generic_ipc *ipc = &skl->ipc;
 	struct skl_ipc_header header = {0};
 	u32 hipcie, hipct, hipcte;
@@ -595,7 +595,7 @@ bool skl_ipc_int_status(struct sst_dsp *ctx)
 			SKL_ADSP_REG_ADSPIS) & SKL_ADSPIS_IPC;
 }
 
-int skl_ipc_init(struct device *dev, struct skl_sst *skl)
+int skl_ipc_init(struct device *dev, struct skl_dev *skl)
 {
 	struct sst_generic_ipc *ipc;
 	int err;
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h
index 9c31a48e99dd..93af08cf41d2 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.h
+++ b/sound/soc/intel/skylake/skl-sst-ipc.h
@@ -10,9 +10,9 @@
 
 #include <linux/irqreturn.h>
 #include "../common/sst-ipc.h"
+#include "skl-sst-dsp.h"
 
 struct sst_dsp;
-struct skl_sst;
 struct sst_generic_ipc;
 
 enum skl_ipc_pipeline_state {
@@ -67,54 +67,6 @@ struct skl_lib_info {
 	const struct firmware *fw;
 };
 
-struct skl_sst {
-	struct device *dev;
-	struct sst_dsp *dsp;
-
-	/* boot */
-	wait_queue_head_t boot_wait;
-	bool boot_complete;
-
-	/* module load */
-	wait_queue_head_t mod_load_wait;
-	bool mod_load_complete;
-	bool mod_load_status;
-
-	/* IPC messaging */
-	struct sst_generic_ipc ipc;
-
-	/* callback for miscbdge */
-	void (*enable_miscbdcge)(struct device *dev, bool enable);
-	/* Is CGCTL.MISCBDCGE disabled */
-	bool miscbdcg_disabled;
-
-	/* Populate module information */
-	struct list_head uuid_list;
-
-	/* Is firmware loaded */
-	bool fw_loaded;
-
-	/* first boot ? */
-	bool is_first_boot;
-
-	/* multi-core */
-	struct skl_dsp_cores cores;
-
-	/* library info */
-	struct skl_lib_info  lib_info[SKL_MAX_LIB];
-	int lib_count;
-
-	/* Callback to update D0i3C register */
-	void (*update_d0i3c)(struct device *dev, bool enable);
-
-	struct skl_d0i3_data d0i3;
-
-	const struct skl_dsp_ops *dsp_ops;
-
-	/* Callback to update dynamic clock and power gating registers */
-	void (*clock_power_gating)(struct device *dev, bool enable);
-};
-
 struct skl_ipc_init_instance_msg {
 	u32 module_id;
 	u32 instance_id;
@@ -204,7 +156,7 @@ void skl_ipc_int_disable(struct sst_dsp *dsp);
 
 bool skl_ipc_int_status(struct sst_dsp *dsp);
 void skl_ipc_free(struct sst_generic_ipc *ipc);
-int skl_ipc_init(struct device *dev, struct skl_sst *skl);
+int skl_ipc_init(struct device *dev, struct skl_dev *skl);
 void skl_clear_module_cnt(struct sst_dsp *ctx);
 
 void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c
index 928c677b506c..d43cbf4a71ef 100644
--- a/sound/soc/intel/skylake/skl-sst-utils.c
+++ b/sound/soc/intel/skylake/skl-sst-utils.c
@@ -8,10 +8,9 @@
 #include <linux/device.h>
 #include <linux/slab.h>
 #include <linux/uuid.h>
-#include "skl-sst-dsp.h"
 #include "../common/sst-dsp.h"
 #include "../common/sst-dsp-priv.h"
-#include "skl-sst-ipc.h"
+#include "skl.h"
 
 #define DEFAULT_HASH_SHA256_LEN 32
 
@@ -99,12 +98,12 @@ static int skl_get_pvtid_map(struct uuid_module *module, int instance_id)
 	return -EINVAL;
 }
 
-int skl_get_pvt_instance_id_map(struct skl_sst *ctx,
+int skl_get_pvt_instance_id_map(struct skl_dev *skl,
 				int module_id, int instance_id)
 {
 	struct uuid_module *module;
 
-	list_for_each_entry(module, &ctx->uuid_list, list) {
+	list_for_each_entry(module, &skl->uuid_list, list) {
 		if (module->id == module_id)
 			return skl_get_pvtid_map(module, instance_id);
 	}
@@ -163,19 +162,19 @@ static inline int skl_pvtid_128(struct uuid_module *module)
 /**
  * skl_get_pvt_id: generate a private id for use as module id
  *
- * @ctx: driver context
+ * @skl: driver context
  * @uuid_mod: module's uuid
  * @instance_id: module's instance id
  *
  * This generates a 128 bit private unique id for a module TYPE so that
  * module instance is unique
  */
-int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id)
+int skl_get_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int instance_id)
 {
 	struct uuid_module *module;
 	int pvt_id;
 
-	list_for_each_entry(module, &ctx->uuid_list, list) {
+	list_for_each_entry(module, &skl->uuid_list, list) {
 		if (guid_equal(uuid_mod, &module->uuid)) {
 
 			pvt_id = skl_pvtid_128(module);
@@ -194,18 +193,18 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id);
 /**
  * skl_put_pvt_id: free up the private id allocated
  *
- * @ctx: driver context
+ * @skl: driver context
  * @uuid_mod: module's uuid
  * @pvt_id: module pvt id
  *
  * This frees a 128 bit private unique id previously generated
  */
-int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id)
+int skl_put_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int *pvt_id)
 {
 	int i;
 	struct uuid_module *module;
 
-	list_for_each_entry(module, &ctx->uuid_list, list) {
+	list_for_each_entry(module, &skl->uuid_list, list) {
 		if (guid_equal(uuid_mod, &module->uuid)) {
 
 			if (*pvt_id != 0)
@@ -234,7 +233,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw,
 	struct adsp_module_entry *mod_entry;
 	int i, num_entry, size;
 	const char *buf;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	struct uuid_module *module;
 	struct firmware stripped_fw;
 	unsigned int safe_file;
@@ -317,11 +316,11 @@ free_uuid_list:
 	return ret;
 }
 
-void skl_freeup_uuid_list(struct skl_sst *ctx)
+void skl_freeup_uuid_list(struct skl_dev *skl)
 {
 	struct uuid_module *uuid, *_uuid;
 
-	list_for_each_entry_safe(uuid, _uuid, &ctx->uuid_list, list) {
+	list_for_each_entry_safe(uuid, _uuid, &skl->uuid_list, list) {
 		list_del(&uuid->list);
 		kfree(uuid);
 	}
@@ -355,16 +354,12 @@ int skl_dsp_strip_extended_manifest(struct firmware *fw)
 }
 
 int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name,
-	struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp,
+	struct skl_dsp_loader_ops dsp_ops, struct skl_dev **dsp,
 	struct sst_dsp_device *skl_dev)
 {
-	struct skl_sst *skl;
+	struct skl_dev *skl = *dsp;
 	struct sst_dsp *sst;
 
-	skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL);
-	if (skl == NULL)
-		return -ENOMEM;
-
 	skl->dev = dev;
 	skl_dev->thread_context = skl;
 	INIT_LIST_HEAD(&skl->uuid_list);
@@ -381,13 +376,11 @@ int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name,
 	INIT_LIST_HEAD(&sst->module_list);
 
 	skl->is_first_boot = true;
-	if (dsp)
-		*dsp = skl;
 
 	return 0;
 }
 
-int skl_prepare_lib_load(struct skl_sst *skl, struct skl_lib_info *linfo,
+int skl_prepare_lib_load(struct skl_dev *skl, struct skl_lib_info *linfo,
 		struct firmware *stripped_fw,
 		unsigned int hdr_offset, int index)
 {
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 70c3a604c381..8af7546def1f 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -16,7 +16,7 @@
 #include "../common/sst-dsp.h"
 #include "../common/sst-dsp-priv.h"
 #include "../common/sst-ipc.h"
-#include "skl-sst-ipc.h"
+#include "skl.h"
 
 #define SKL_BASEFW_TIMEOUT	300
 #define SKL_INIT_TIMEOUT	1000
@@ -66,7 +66,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx,
 static int skl_load_base_firmware(struct sst_dsp *ctx)
 {
 	int ret = 0, i;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	struct firmware stripped_fw;
 	u32 reg;
 
@@ -161,7 +161,7 @@ static int skl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id)
 {
 	int ret;
 	struct skl_ipc_dxstate_info dx;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
 
 	/* If core0 is being turned on, we need to load the FW */
@@ -215,7 +215,7 @@ static int skl_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id)
 {
 	int ret;
 	struct skl_ipc_dxstate_info dx;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	unsigned int core_mask = SKL_DSP_CORE_MASK(core_id);
 
 	dx.core_mask = core_mask;
@@ -332,7 +332,7 @@ static int skl_transfer_module(struct sst_dsp *ctx, const void *data,
 			u32 size, u16 mod_id, u8 table_id, bool is_module)
 {
 	int ret, bytes_left, curr_pos;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	skl->mod_load_complete = false;
 
 	bytes_left = ctx->cl_dev.ops.cl_copy_to_dmabuf(ctx, data, size, false);
@@ -384,7 +384,7 @@ out:
 static int
 skl_load_library(struct sst_dsp *ctx, struct skl_lib_info *linfo, int lib_count)
 {
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	struct firmware stripped_fw;
 	int ret, i;
 
@@ -443,7 +443,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid)
 static int skl_unload_module(struct sst_dsp *ctx, u16 mod_id)
 {
 	int usage_cnt;
-	struct skl_sst *skl = ctx->thread_context;
+	struct skl_dev *skl = ctx->thread_context;
 	int ret = 0;
 
 	usage_cnt = skl_put_module(ctx, mod_id);
@@ -518,9 +518,10 @@ static struct sst_dsp_device skl_dev = {
 };
 
 int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
-		const char *fw_name, struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp)
+		const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+		struct skl_dev **dsp)
 {
-	struct skl_sst *skl;
+	struct skl_dev *skl;
 	struct sst_dsp *sst;
 	int ret;
 
@@ -554,10 +555,10 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 }
 EXPORT_SYMBOL_GPL(skl_sst_dsp_init);
 
-int skl_sst_init_fw(struct device *dev, struct skl_sst *ctx)
+int skl_sst_init_fw(struct device *dev, struct skl_dev *skl)
 {
 	int ret;
-	struct sst_dsp *sst = ctx->dsp;
+	struct sst_dsp *sst = skl->dsp;
 
 	ret = sst->fw_ops.load_fw(sst);
 	if (ret < 0) {
@@ -567,32 +568,32 @@ int skl_sst_init_fw(struct device *dev, struct skl_sst *ctx)
 
 	skl_dsp_init_core_state(sst);
 
-	if (ctx->lib_count > 1) {
-		ret = sst->fw_ops.load_library(sst, ctx->lib_info,
-						ctx->lib_count);
+	if (skl->lib_count > 1) {
+		ret = sst->fw_ops.load_library(sst, skl->lib_info,
+						skl->lib_count);
 		if (ret < 0) {
 			dev_err(dev, "Load Library failed : %x\n", ret);
 			return ret;
 		}
 	}
-	ctx->is_first_boot = false;
+	skl->is_first_boot = false;
 
 	return 0;
 }
 EXPORT_SYMBOL_GPL(skl_sst_init_fw);
 
-void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
+void skl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl)
 {
 
-	if (ctx->dsp->fw)
-		release_firmware(ctx->dsp->fw);
-	skl_clear_module_table(ctx->dsp);
-	skl_freeup_uuid_list(ctx);
-	skl_ipc_free(&ctx->ipc);
-	ctx->dsp->ops->free(ctx->dsp);
-	if (ctx->boot_complete) {
-		ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
-		skl_cldma_int_disable(ctx->dsp);
+	if (skl->dsp->fw)
+		release_firmware(skl->dsp->fw);
+	skl_clear_module_table(skl->dsp);
+	skl_freeup_uuid_list(skl);
+	skl_ipc_free(&skl->ipc);
+	skl->dsp->ops->free(skl->dsp);
+	if (skl->boot_complete) {
+		skl->dsp->cl_dev.ops.cl_cleanup_controller(skl->dsp);
+		skl_cldma_int_disable(skl->dsp);
 	}
 }
 EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup);
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 6241e35213af..abf860109607 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -45,9 +45,9 @@ static const int mic_quatro_list[][SKL_CH_QUATRO] = {
 #define CHECK_HW_PARAMS(ch, freq, bps, prm_ch, prm_freq, prm_bps) \
 	((ch == prm_ch) && (bps == prm_bps) && (freq == prm_freq))
 
-void skl_tplg_d0i3_get(struct skl *skl, enum d0i3_capability caps)
+void skl_tplg_d0i3_get(struct skl_dev *skl, enum d0i3_capability caps)
 {
-	struct skl_d0i3_data *d0i3 =  &skl->skl_sst->d0i3;
+	struct skl_d0i3_data *d0i3 =  &skl->d0i3;
 
 	switch (caps) {
 	case SKL_D0I3_NONE:
@@ -64,9 +64,9 @@ void skl_tplg_d0i3_get(struct skl *skl, enum d0i3_capability caps)
 	}
 }
 
-void skl_tplg_d0i3_put(struct skl *skl, enum d0i3_capability caps)
+void skl_tplg_d0i3_put(struct skl_dev *skl, enum d0i3_capability caps)
 {
-	struct skl_d0i3_data *d0i3 =  &skl->skl_sst->d0i3;
+	struct skl_d0i3_data *d0i3 =  &skl->d0i3;
 
 	switch (caps) {
 	case SKL_D0I3_NONE:
@@ -113,18 +113,16 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w,
  * Each pipelines needs memory to be allocated. Check if we have free memory
  * from available pool.
  */
-static bool skl_is_pipe_mem_avail(struct skl *skl,
+static bool skl_is_pipe_mem_avail(struct skl_dev *skl,
 				struct skl_module_cfg *mconfig)
 {
-	struct skl_sst *ctx = skl->skl_sst;
-
 	if (skl->resource.mem + mconfig->pipe->memory_pages >
 				skl->resource.max_mem) {
-		dev_err(ctx->dev,
+		dev_err(skl->dev,
 				"%s: module_id %d instance %d\n", __func__,
 				mconfig->id.module_id,
 				mconfig->id.instance_id);
-		dev_err(ctx->dev,
+		dev_err(skl->dev,
 				"exceeds ppl memory available %d mem %d\n",
 				skl->resource.max_mem, skl->resource.mem);
 		return false;
@@ -138,7 +136,7 @@ static bool skl_is_pipe_mem_avail(struct skl *skl,
  * Note: DSP does actual memory management we only keep track for complete
  * pool
  */
-static void skl_tplg_alloc_pipe_mem(struct skl *skl,
+static void skl_tplg_alloc_pipe_mem(struct skl_dev *skl,
 				struct skl_module_cfg *mconfig)
 {
 	skl->resource.mem += mconfig->pipe->memory_pages;
@@ -152,18 +150,17 @@ static void skl_tplg_alloc_pipe_mem(struct skl *skl,
  * pipe.
  */
 
-static bool skl_is_pipe_mcps_avail(struct skl *skl,
+static bool skl_is_pipe_mcps_avail(struct skl_dev *skl,
 				struct skl_module_cfg *mconfig)
 {
-	struct skl_sst *ctx = skl->skl_sst;
 	u8 res_idx = mconfig->res_idx;
 	struct skl_module_res *res = &mconfig->module->resources[res_idx];
 
 	if (skl->resource.mcps + res->cps > skl->resource.max_mcps) {
-		dev_err(ctx->dev,
+		dev_err(skl->dev,
 			"%s: module_id %d instance %d\n", __func__,
 			mconfig->id.module_id, mconfig->id.instance_id);
-		dev_err(ctx->dev,
+		dev_err(skl->dev,
 			"exceeds ppl mcps available %d > mem %d\n",
 			skl->resource.max_mcps, skl->resource.mcps);
 		return false;
@@ -172,7 +169,7 @@ static bool skl_is_pipe_mcps_avail(struct skl *skl,
 	}
 }
 
-static void skl_tplg_alloc_pipe_mcps(struct skl *skl,
+static void skl_tplg_alloc_pipe_mcps(struct skl_dev *skl,
 				struct skl_module_cfg *mconfig)
 {
 	u8 res_idx = mconfig->res_idx;
@@ -185,7 +182,7 @@ static void skl_tplg_alloc_pipe_mcps(struct skl *skl,
  * Free the mcps when tearing down
  */
 static void
-skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig)
+skl_tplg_free_pipe_mcps(struct skl_dev *skl, struct skl_module_cfg *mconfig)
 {
 	u8 res_idx = mconfig->res_idx;
 	struct skl_module_res *res = &mconfig->module->resources[res_idx];
@@ -197,30 +194,29 @@ skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig)
  * Free the memory when tearing down
  */
 static void
-skl_tplg_free_pipe_mem(struct skl *skl, struct skl_module_cfg *mconfig)
+skl_tplg_free_pipe_mem(struct skl_dev *skl, struct skl_module_cfg *mconfig)
 {
 	skl->resource.mem -= mconfig->pipe->memory_pages;
 }
 
 
-static void skl_dump_mconfig(struct skl_sst *ctx,
-					struct skl_module_cfg *mcfg)
+static void skl_dump_mconfig(struct skl_dev *skl, struct skl_module_cfg *mcfg)
 {
 	struct skl_module_iface *iface = &mcfg->module->formats[0];
 
-	dev_dbg(ctx->dev, "Dumping config\n");
-	dev_dbg(ctx->dev, "Input Format:\n");
-	dev_dbg(ctx->dev, "channels = %d\n", iface->inputs[0].fmt.channels);
-	dev_dbg(ctx->dev, "s_freq = %d\n", iface->inputs[0].fmt.s_freq);
-	dev_dbg(ctx->dev, "ch_cfg = %d\n", iface->inputs[0].fmt.ch_cfg);
-	dev_dbg(ctx->dev, "valid bit depth = %d\n",
+	dev_dbg(skl->dev, "Dumping config\n");
+	dev_dbg(skl->dev, "Input Format:\n");
+	dev_dbg(skl->dev, "channels = %d\n", iface->inputs[0].fmt.channels);
+	dev_dbg(skl->dev, "s_freq = %d\n", iface->inputs[0].fmt.s_freq);
+	dev_dbg(skl->dev, "ch_cfg = %d\n", iface->inputs[0].fmt.ch_cfg);
+	dev_dbg(skl->dev, "valid bit depth = %d\n",
 				iface->inputs[0].fmt.valid_bit_depth);
-	dev_dbg(ctx->dev, "Output Format:\n");
-	dev_dbg(ctx->dev, "channels = %d\n", iface->outputs[0].fmt.channels);
-	dev_dbg(ctx->dev, "s_freq = %d\n", iface->outputs[0].fmt.s_freq);
-	dev_dbg(ctx->dev, "valid bit depth = %d\n",
+	dev_dbg(skl->dev, "Output Format:\n");
+	dev_dbg(skl->dev, "channels = %d\n", iface->outputs[0].fmt.channels);
+	dev_dbg(skl->dev, "s_freq = %d\n", iface->outputs[0].fmt.s_freq);
+	dev_dbg(skl->dev, "valid bit depth = %d\n",
 				iface->outputs[0].fmt.valid_bit_depth);
-	dev_dbg(ctx->dev, "ch_cfg = %d\n", iface->outputs[0].fmt.ch_cfg);
+	dev_dbg(skl->dev, "ch_cfg = %d\n", iface->outputs[0].fmt.ch_cfg);
 }
 
 static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs)
@@ -322,7 +318,7 @@ static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg,
  * params, so once we have calculate params, we need buffer calculation as
  * well.
  */
-static void skl_tplg_update_buffer_size(struct skl_sst *ctx,
+static void skl_tplg_update_buffer_size(struct skl_dev *skl,
 				struct skl_module_cfg *mcfg)
 {
 	int multiplier = 1;
@@ -374,13 +370,12 @@ static u8 skl_tplg_be_dev_type(int dev_type)
 }
 
 static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w,
-						struct skl_sst *ctx)
+						struct skl_dev *skl)
 {
 	struct skl_module_cfg *m_cfg = w->priv;
 	int link_type, dir;
 	u32 ch, s_freq, s_fmt;
 	struct nhlt_specific_cfg *cfg;
-	struct skl *skl = get_skl_ctx(ctx->dev);
 	u8 dev_type = skl_tplg_be_dev_type(m_cfg->dev_type);
 	int fmt_idx = m_cfg->fmt_idx;
 	struct skl_module_iface *m_iface = &m_cfg->module->formats[fmt_idx];
@@ -389,7 +384,7 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w,
 	if (m_cfg->formats_config.caps_size > 0)
 		return 0;
 
-	dev_dbg(ctx->dev, "Applying default cfg blob\n");
+	dev_dbg(skl->dev, "Applying default cfg blob\n");
 	switch (m_cfg->dev_type) {
 	case SKL_DEVICE_DMIC:
 		link_type = NHLT_LINK_DMIC;
@@ -425,9 +420,9 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w,
 		m_cfg->formats_config.caps_size = cfg->size;
 		m_cfg->formats_config.caps = (u32 *) &cfg->caps;
 	} else {
-		dev_err(ctx->dev, "Blob NULL for id %x type %d dirn %d\n",
+		dev_err(skl->dev, "Blob NULL for id %x type %d dirn %d\n",
 					m_cfg->vbus_id, link_type, dir);
-		dev_err(ctx->dev, "PCM: ch %d, freq %d, fmt %d\n",
+		dev_err(skl->dev, "PCM: ch %d, freq %d, fmt %d\n",
 					ch, s_freq, s_fmt);
 		return -EIO;
 	}
@@ -436,7 +431,7 @@ static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w,
 }
 
 static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w,
-							struct skl_sst *ctx)
+							struct skl_dev *skl)
 {
 	struct skl_module_cfg *m_cfg = w->priv;
 	struct skl_pipe_params *params = m_cfg->pipe->p_params;
@@ -446,10 +441,10 @@ static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w,
 	if (!m_cfg->params_fixup)
 		return;
 
-	dev_dbg(ctx->dev, "Mconfig for widget=%s BEFORE updation\n",
+	dev_dbg(skl->dev, "Mconfig for widget=%s BEFORE updation\n",
 				w->name);
 
-	skl_dump_mconfig(ctx, m_cfg);
+	skl_dump_mconfig(skl, m_cfg);
 
 	if (p_conn_type == SKL_PIPE_CONN_TYPE_FE)
 		is_fe = true;
@@ -457,12 +452,12 @@ static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w,
 		is_fe = false;
 
 	skl_tplg_update_params_fixup(m_cfg, params, is_fe);
-	skl_tplg_update_buffer_size(ctx, m_cfg);
+	skl_tplg_update_buffer_size(skl, m_cfg);
 
-	dev_dbg(ctx->dev, "Mconfig for widget=%s AFTER updation\n",
+	dev_dbg(skl->dev, "Mconfig for widget=%s AFTER updation\n",
 				w->name);
 
-	skl_dump_mconfig(ctx, m_cfg);
+	skl_dump_mconfig(skl, m_cfg);
 }
 
 /*
@@ -471,7 +466,7 @@ static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w,
  * set module params will be done after module is initialised.
  */
 static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w,
-						struct skl_sst *ctx)
+						struct skl_dev *skl)
 {
 	int i, ret;
 	struct skl_module_cfg *mconfig = w->priv;
@@ -483,7 +478,7 @@ static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w,
 	if (mconfig->formats_config.caps_size > 0 &&
 		mconfig->formats_config.set_params == SKL_PARAM_SET) {
 		sp_cfg = &mconfig->formats_config;
-		ret = skl_set_module_params(ctx, sp_cfg->caps,
+		ret = skl_set_module_params(skl, sp_cfg->caps,
 					sp_cfg->caps_size,
 					sp_cfg->param_id, mconfig);
 		if (ret < 0)
@@ -497,7 +492,7 @@ static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w,
 			bc = (struct skl_algo_data *)sb->dobj.private;
 
 			if (bc->set_params == SKL_PARAM_SET) {
-				ret = skl_set_module_params(ctx,
+				ret = skl_set_module_params(skl,
 						(u32 *)bc->params, bc->size,
 						bc->param_id, mconfig);
 				if (ret < 0)
@@ -542,15 +537,15 @@ static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w)
 	return 0;
 }
 
-static int skl_tplg_module_prepare(struct skl_sst *ctx, struct skl_pipe *pipe,
+static int skl_tplg_module_prepare(struct skl_dev *skl, struct skl_pipe *pipe,
 		struct snd_soc_dapm_widget *w, struct skl_module_cfg *mcfg)
 {
 	switch (mcfg->dev_type) {
 	case SKL_DEVICE_HDAHOST:
-		return skl_pcm_host_dma_prepare(ctx->dev, pipe->p_params);
+		return skl_pcm_host_dma_prepare(skl->dev, pipe->p_params);
 
 	case SKL_DEVICE_HDALINK:
-		return skl_pcm_link_dma_prepare(ctx->dev, pipe->p_params);
+		return skl_pcm_link_dma_prepare(skl->dev, pipe->p_params);
 	}
 
 	return 0;
@@ -562,12 +557,11 @@ static int skl_tplg_module_prepare(struct skl_sst *ctx, struct skl_pipe *pipe,
  * skl_init_module() routine, so invoke that for all modules in a pipeline
  */
 static int
-skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
+skl_tplg_init_pipe_modules(struct skl_dev *skl, struct skl_pipe *pipe)
 {
 	struct skl_pipe_module *w_module;
 	struct snd_soc_dapm_widget *w;
 	struct skl_module_cfg *mconfig;
-	struct skl_sst *ctx = skl->skl_sst;
 	u8 cfg_idx;
 	int ret = 0;
 
@@ -578,7 +572,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 
 		/* check if module ids are populated */
 		if (mconfig->id.module_id < 0) {
-			dev_err(skl->skl_sst->dev,
+			dev_err(skl->dev,
 					"module %pUL id not populated\n",
 					(guid_t *)mconfig->guid);
 			return -EIO;
@@ -592,8 +586,8 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 		if (!skl_is_pipe_mcps_avail(skl, mconfig))
 			return -ENOMEM;
 
-		if (mconfig->module->loadable && ctx->dsp->fw_ops.load_mod) {
-			ret = ctx->dsp->fw_ops.load_mod(ctx->dsp,
+		if (mconfig->module->loadable && skl->dsp->fw_ops.load_mod) {
+			ret = skl->dsp->fw_ops.load_mod(skl->dsp,
 				mconfig->id.module_id, mconfig->guid);
 			if (ret < 0)
 				return ret;
@@ -602,50 +596,50 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 		}
 
 		/* prepare the DMA if the module is gateway cpr */
-		ret = skl_tplg_module_prepare(ctx, pipe, w, mconfig);
+		ret = skl_tplg_module_prepare(skl, pipe, w, mconfig);
 		if (ret < 0)
 			return ret;
 
 		/* update blob if blob is null for be with default value */
-		skl_tplg_update_be_blob(w, ctx);
+		skl_tplg_update_be_blob(w, skl);
 
 		/*
 		 * apply fix/conversion to module params based on
 		 * FE/BE params
 		 */
-		skl_tplg_update_module_params(w, ctx);
+		skl_tplg_update_module_params(w, skl);
 		uuid_mod = (guid_t *)mconfig->guid;
-		mconfig->id.pvt_id = skl_get_pvt_id(ctx, uuid_mod,
+		mconfig->id.pvt_id = skl_get_pvt_id(skl, uuid_mod,
 						mconfig->id.instance_id);
 		if (mconfig->id.pvt_id < 0)
 			return ret;
 		skl_tplg_set_module_init_data(w);
 
-		ret = skl_dsp_get_core(ctx->dsp, mconfig->core_id);
+		ret = skl_dsp_get_core(skl->dsp, mconfig->core_id);
 		if (ret < 0) {
-			dev_err(ctx->dev, "Failed to wake up core %d ret=%d\n",
+			dev_err(skl->dev, "Failed to wake up core %d ret=%d\n",
 						mconfig->core_id, ret);
 			return ret;
 		}
 
-		ret = skl_init_module(ctx, mconfig);
+		ret = skl_init_module(skl, mconfig);
 		if (ret < 0) {
-			skl_put_pvt_id(ctx, uuid_mod, &mconfig->id.pvt_id);
+			skl_put_pvt_id(skl, uuid_mod, &mconfig->id.pvt_id);
 			goto err;
 		}
 		skl_tplg_alloc_pipe_mcps(skl, mconfig);
-		ret = skl_tplg_set_module_params(w, ctx);
+		ret = skl_tplg_set_module_params(w, skl);
 		if (ret < 0)
 			goto err;
 	}
 
 	return 0;
 err:
-	skl_dsp_put_core(ctx->dsp, mconfig->core_id);
+	skl_dsp_put_core(skl->dsp, mconfig->core_id);
 	return ret;
 }
 
-static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
+static int skl_tplg_unload_pipe_modules(struct skl_dev *skl,
 	 struct skl_pipe *pipe)
 {
 	int ret = 0;
@@ -657,19 +651,19 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
 		mconfig  = w_module->w->priv;
 		uuid_mod = (guid_t *)mconfig->guid;
 
-		if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod &&
+		if (mconfig->module->loadable && skl->dsp->fw_ops.unload_mod &&
 			mconfig->m_state > SKL_MODULE_UNINIT) {
-			ret = ctx->dsp->fw_ops.unload_mod(ctx->dsp,
+			ret = skl->dsp->fw_ops.unload_mod(skl->dsp,
 						mconfig->id.module_id);
 			if (ret < 0)
 				return -EIO;
 		}
-		skl_put_pvt_id(ctx, uuid_mod, &mconfig->id.pvt_id);
+		skl_put_pvt_id(skl, uuid_mod, &mconfig->id.pvt_id);
 
-		ret = skl_dsp_put_core(ctx->dsp, mconfig->core_id);
+		ret = skl_dsp_put_core(skl->dsp, mconfig->core_id);
 		if (ret < 0) {
 			/* don't return; continue with other modules */
-			dev_err(ctx->dev, "Failed to sleep core %d ret=%d\n",
+			dev_err(skl->dev, "Failed to sleep core %d ret=%d\n",
 				mconfig->core_id, ret);
 		}
 	}
@@ -686,9 +680,8 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
  * 0th configuratation by default for such pipes.
  */
 static int
-skl_tplg_get_pipe_config(struct skl *skl, struct skl_module_cfg *mconfig)
+skl_tplg_get_pipe_config(struct skl_dev *skl, struct skl_module_cfg *mconfig)
 {
-	struct skl_sst *ctx = skl->skl_sst;
 	struct skl_pipe *pipe = mconfig->pipe;
 	struct skl_pipe_params *params = pipe->p_params;
 	struct skl_path_config *pconfig = &pipe->configs[0];
@@ -702,7 +695,7 @@ skl_tplg_get_pipe_config(struct skl *skl, struct skl_module_cfg *mconfig)
 	}
 
 	if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE) {
-		dev_dbg(ctx->dev, "No conn_type detected, take 0th config\n");
+		dev_dbg(skl->dev, "No conn_type detected, take 0th config\n");
 		pipe->cur_config_idx = 0;
 		pipe->memory_pages = pconfig->mem_pages;
 
@@ -726,13 +719,13 @@ skl_tplg_get_pipe_config(struct skl *skl, struct skl_module_cfg *mconfig)
 				    fmt->channels, fmt->freq, fmt->bps)) {
 			pipe->cur_config_idx = i;
 			pipe->memory_pages = pconfig->mem_pages;
-			dev_dbg(ctx->dev, "Using pipe config: %d\n", i);
+			dev_dbg(skl->dev, "Using pipe config: %d\n", i);
 
 			return 0;
 		}
 	}
 
-	dev_err(ctx->dev, "Invalid pipe config: %d %d %d for pipe: %d\n",
+	dev_err(skl->dev, "Invalid pipe config: %d %d %d for pipe: %d\n",
 		params->ch, params->s_freq, params->s_fmt, pipe->ppl_id);
 	return -EINVAL;
 }
@@ -746,14 +739,13 @@ skl_tplg_get_pipe_config(struct skl *skl, struct skl_module_cfg *mconfig)
  *   - finally bind all modules together
  */
 static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
-							struct skl *skl)
+							struct skl_dev *skl)
 {
 	int ret;
 	struct skl_module_cfg *mconfig = w->priv;
 	struct skl_pipe_module *w_module;
 	struct skl_pipe *s_pipe = mconfig->pipe;
 	struct skl_module_cfg *src_module = NULL, *dst_module, *module;
-	struct skl_sst *ctx = skl->skl_sst;
 	struct skl_module_deferred_bind *modules;
 
 	ret = skl_tplg_get_pipe_config(skl, mconfig);
@@ -771,7 +763,7 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	 * Create a list of modules for pipe.
 	 * This list contains modules from source to sink
 	 */
-	ret = skl_create_pipeline(ctx, mconfig->pipe);
+	ret = skl_create_pipeline(skl, mconfig->pipe);
 	if (ret < 0)
 		return ret;
 
@@ -792,7 +784,7 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 			continue;
 		}
 
-		ret = skl_bind_modules(ctx, src_module, dst_module);
+		ret = skl_bind_modules(skl, src_module, dst_module);
 		if (ret < 0)
 			return ret;
 
@@ -810,7 +802,7 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 		list_for_each_entry(modules, &skl->bind_list, node) {
 			module = w_module->w->priv;
 			if (modules->dst == module)
-				skl_bind_modules(ctx, modules->src,
+				skl_bind_modules(skl, modules->src,
 							modules->dst);
 		}
 	}
@@ -818,7 +810,7 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int skl_fill_sink_instance_id(struct skl_sst *ctx, u32 *params,
+static int skl_fill_sink_instance_id(struct skl_dev *skl, u32 *params,
 				int size, struct skl_module_cfg *mcfg)
 {
 	int i, pvt_id;
@@ -829,7 +821,7 @@ static int skl_fill_sink_instance_id(struct skl_sst *ctx, u32 *params,
 		struct skl_mod_inst_map *inst = kpb_params->u.map;
 
 		for (i = 0; i < kpb_params->num_modules; i++) {
-			pvt_id = skl_get_pvt_instance_id_map(ctx, inst->mod_id,
+			pvt_id = skl_get_pvt_instance_id_map(skl, inst->mod_id,
 								inst->inst_id);
 			if (pvt_id < 0)
 				return -EINVAL;
@@ -849,7 +841,7 @@ static int skl_fill_sink_instance_id(struct skl_sst *ctx, u32 *params,
  * send params after binding
  */
 static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
-			struct skl_module_cfg *mcfg, struct skl_sst *ctx)
+			struct skl_module_cfg *mcfg, struct skl_dev *skl)
 {
 	int i, ret;
 	struct skl_module_cfg *mconfig = w->priv;
@@ -876,7 +868,7 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
 	if (mconfig->formats_config.caps_size > 0 &&
 		mconfig->formats_config.set_params == SKL_PARAM_BIND) {
 		sp_cfg = &mconfig->formats_config;
-		ret = skl_set_module_params(ctx, sp_cfg->caps,
+		ret = skl_set_module_params(skl, sp_cfg->caps,
 					sp_cfg->caps_size,
 					sp_cfg->param_id, mconfig);
 		if (ret < 0)
@@ -894,10 +886,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
 				if (!params)
 					return -ENOMEM;
 
-				skl_fill_sink_instance_id(ctx, params, bc->max,
+				skl_fill_sink_instance_id(skl, params, bc->max,
 								mconfig);
 
-				ret = skl_set_module_params(ctx, params,
+				ret = skl_set_module_params(skl, params,
 						bc->max, bc->param_id, mconfig);
 				kfree(params);
 
@@ -910,11 +902,11 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid)
+static int skl_get_module_id(struct skl_dev *skl, guid_t *uuid)
 {
 	struct uuid_module *module;
 
-	list_for_each_entry(module, &ctx->uuid_list, list) {
+	list_for_each_entry(module, &skl->uuid_list, list) {
 		if (guid_equal(uuid, &module->uuid))
 			return module->id;
 	}
@@ -922,7 +914,7 @@ static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid)
 	return -EINVAL;
 }
 
-static int skl_tplg_find_moduleid_from_uuid(struct skl *skl,
+static int skl_tplg_find_moduleid_from_uuid(struct skl_dev *skl,
 					const struct snd_kcontrol_new *k)
 {
 	struct soc_bytes_ext *sb = (void *) k->private_value;
@@ -942,7 +934,7 @@ static int skl_tplg_find_moduleid_from_uuid(struct skl *skl,
 		params->num_modules = uuid_params->num_modules;
 
 		for (i = 0; i < uuid_params->num_modules; i++) {
-			module_id = skl_get_module_id(skl->skl_sst,
+			module_id = skl_get_module_id(skl,
 				&uuid_params->u.map_uuid[i].mod_uuid);
 			if (module_id < 0) {
 				devm_kfree(bus->dev, params);
@@ -966,7 +958,7 @@ static int skl_tplg_find_moduleid_from_uuid(struct skl *skl,
  * Retrieve the module id from UUID mentioned in the
  * post bind params
  */
-void skl_tplg_add_moduleid_in_bind_params(struct skl *skl,
+void skl_tplg_add_moduleid_in_bind_params(struct skl_dev *skl,
 				struct snd_soc_dapm_widget *w)
 {
 	struct skl_module_cfg *mconfig = w->priv;
@@ -985,12 +977,12 @@ void skl_tplg_add_moduleid_in_bind_params(struct skl *skl,
 			SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) &&
 			(skl_tplg_find_moduleid_from_uuid(skl,
 			&w->kcontrol_news[i]) < 0))
-			dev_err(skl->skl_sst->dev,
+			dev_err(skl->dev,
 				"%s: invalid kpb post bind params\n",
 				__func__);
 }
 
-static int skl_tplg_module_add_deferred_bind(struct skl *skl,
+static int skl_tplg_module_add_deferred_bind(struct skl_dev *skl,
 	struct skl_module_cfg *src, struct skl_module_cfg *dst)
 {
 	struct skl_module_deferred_bind *m_list, *modules;
@@ -1028,26 +1020,27 @@ static int skl_tplg_module_add_deferred_bind(struct skl *skl,
 }
 
 static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
-				struct skl *skl,
+				struct skl_dev *skl,
 				struct snd_soc_dapm_widget *src_w,
 				struct skl_module_cfg *src_mconfig)
 {
 	struct snd_soc_dapm_path *p;
 	struct snd_soc_dapm_widget *sink = NULL, *next_sink = NULL;
 	struct skl_module_cfg *sink_mconfig;
-	struct skl_sst *ctx = skl->skl_sst;
 	int ret;
 
 	snd_soc_dapm_widget_for_each_sink_path(w, p) {
 		if (!p->connect)
 			continue;
 
-		dev_dbg(ctx->dev, "%s: src widget=%s\n", __func__, w->name);
-		dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name);
+		dev_dbg(skl->dev,
+			"%s: src widget=%s\n", __func__, w->name);
+		dev_dbg(skl->dev,
+			"%s: sink widget=%s\n", __func__, p->sink->name);
 
 		next_sink = p->sink;
 
-		if (!is_skl_dsp_widget_type(p->sink, ctx->dev))
+		if (!is_skl_dsp_widget_type(p->sink, skl->dev))
 			return skl_tplg_bind_sinks(p->sink, skl, src_w, src_mconfig);
 
 		/*
@@ -1056,7 +1049,7 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
 		 * they are ones used for SKL so check that first
 		 */
 		if ((p->sink->priv != NULL) &&
-				is_skl_dsp_widget_type(p->sink, ctx->dev)) {
+				is_skl_dsp_widget_type(p->sink, skl->dev)) {
 
 			sink = p->sink;
 			sink_mconfig = sink->priv;
@@ -1088,19 +1081,21 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
 				continue;
 
 			/* Bind source to sink, mixin is always source */
-			ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig);
+			ret = skl_bind_modules(skl, src_mconfig, sink_mconfig);
 			if (ret)
 				return ret;
 
 			/* set module params after bind */
-			skl_tplg_set_module_bind_params(src_w, src_mconfig, ctx);
-			skl_tplg_set_module_bind_params(sink, sink_mconfig, ctx);
+			skl_tplg_set_module_bind_params(src_w,
+					src_mconfig, skl);
+			skl_tplg_set_module_bind_params(sink,
+					sink_mconfig, skl);
 
 			/* Start sinks pipe first */
 			if (sink_mconfig->pipe->state != SKL_PIPE_STARTED) {
 				if (sink_mconfig->pipe->conn_type !=
 							SKL_PIPE_CONN_TYPE_FE)
-					ret = skl_run_pipe(ctx,
+					ret = skl_run_pipe(skl,
 							sink_mconfig->pipe);
 				if (ret)
 					return ret;
@@ -1125,10 +1120,9 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
  *   - Then run current pipe
  */
 static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
-								struct skl *skl)
+							struct skl_dev *skl)
 {
 	struct skl_module_cfg *src_mconfig;
-	struct skl_sst *ctx = skl->skl_sst;
 	int ret = 0;
 
 	src_mconfig = w->priv;
@@ -1144,25 +1138,24 @@ static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 
 	/* Start source pipe last after starting all sinks */
 	if (src_mconfig->pipe->conn_type != SKL_PIPE_CONN_TYPE_FE)
-		return skl_run_pipe(ctx, src_mconfig->pipe);
+		return skl_run_pipe(skl, src_mconfig->pipe);
 
 	return 0;
 }
 
 static struct snd_soc_dapm_widget *skl_get_src_dsp_widget(
-		struct snd_soc_dapm_widget *w, struct skl *skl)
+		struct snd_soc_dapm_widget *w, struct skl_dev *skl)
 {
 	struct snd_soc_dapm_path *p;
 	struct snd_soc_dapm_widget *src_w = NULL;
-	struct skl_sst *ctx = skl->skl_sst;
 
 	snd_soc_dapm_widget_for_each_source_path(w, p) {
 		src_w = p->source;
 		if (!p->connect)
 			continue;
 
-		dev_dbg(ctx->dev, "sink widget=%s\n", w->name);
-		dev_dbg(ctx->dev, "src widget=%s\n", p->source->name);
+		dev_dbg(skl->dev, "sink widget=%s\n", w->name);
+		dev_dbg(skl->dev, "src widget=%s\n", p->source->name);
 
 		/*
 		 * here we will check widgets in sink pipelines, so that can
@@ -1170,7 +1163,7 @@ static struct snd_soc_dapm_widget *skl_get_src_dsp_widget(
 		 * ones used for SKL so check that first
 		 */
 		if ((p->source->priv != NULL) &&
-				is_skl_dsp_widget_type(p->source, ctx->dev)) {
+				is_skl_dsp_widget_type(p->source, skl->dev)) {
 			return p->source;
 		}
 	}
@@ -1191,12 +1184,11 @@ static struct snd_soc_dapm_widget *skl_get_src_dsp_widget(
  *	- start this pipeline
  */
 static int skl_tplg_mixer_dapm_post_pmu_event(struct snd_soc_dapm_widget *w,
-							struct skl *skl)
+							struct skl_dev *skl)
 {
 	int ret = 0;
 	struct snd_soc_dapm_widget *source, *sink;
 	struct skl_module_cfg *src_mconfig, *sink_mconfig;
-	struct skl_sst *ctx = skl->skl_sst;
 	int src_pipe_started = 0;
 
 	sink = w;
@@ -1222,16 +1214,16 @@ static int skl_tplg_mixer_dapm_post_pmu_event(struct snd_soc_dapm_widget *w,
 	}
 
 	if (src_pipe_started) {
-		ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig);
+		ret = skl_bind_modules(skl, src_mconfig, sink_mconfig);
 		if (ret)
 			return ret;
 
 		/* set module params after bind */
-		skl_tplg_set_module_bind_params(source, src_mconfig, ctx);
-		skl_tplg_set_module_bind_params(sink, sink_mconfig, ctx);
+		skl_tplg_set_module_bind_params(source, src_mconfig, skl);
+		skl_tplg_set_module_bind_params(sink, sink_mconfig, skl);
 
 		if (sink_mconfig->pipe->conn_type != SKL_PIPE_CONN_TYPE_FE)
-			ret = skl_run_pipe(ctx, sink_mconfig->pipe);
+			ret = skl_run_pipe(skl, sink_mconfig->pipe);
 	}
 
 	return ret;
@@ -1244,16 +1236,15 @@ static int skl_tplg_mixer_dapm_post_pmu_event(struct snd_soc_dapm_widget *w,
  *   - unbind with source pipelines if still connected
  */
 static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
-							struct skl *skl)
+							struct skl_dev *skl)
 {
 	struct skl_module_cfg *src_mconfig, *sink_mconfig;
 	int ret = 0, i;
-	struct skl_sst *ctx = skl->skl_sst;
 
 	sink_mconfig = w->priv;
 
 	/* Stop the pipe */
-	ret = skl_stop_pipe(ctx, sink_mconfig->pipe);
+	ret = skl_stop_pipe(skl, sink_mconfig->pipe);
 	if (ret)
 		return ret;
 
@@ -1263,7 +1254,7 @@ static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
 			if (!src_mconfig)
 				continue;
 
-			ret = skl_unbind_modules(ctx,
+			ret = skl_unbind_modules(skl,
 						src_mconfig, sink_mconfig);
 		}
 	}
@@ -1280,12 +1271,11 @@ static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
  *     deleted, pipeline delete is enough here
  */
 static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
-							struct skl *skl)
+							struct skl_dev *skl)
 {
 	struct skl_module_cfg *mconfig = w->priv;
 	struct skl_pipe_module *w_module;
 	struct skl_module_cfg *src_module = NULL, *dst_module;
-	struct skl_sst *ctx = skl->skl_sst;
 	struct skl_pipe *s_pipe = mconfig->pipe;
 	struct skl_module_deferred_bind *modules, *tmp;
 
@@ -1307,7 +1297,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 			 * modules from deferred bind list.
 			 */
 			if (modules->dst == src_module) {
-				skl_unbind_modules(ctx, modules->src,
+				skl_unbind_modules(skl, modules->src,
 						modules->dst);
 			}
 
@@ -1334,18 +1324,18 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 			continue;
 		}
 
-		skl_unbind_modules(ctx, src_module, dst_module);
+		skl_unbind_modules(skl, src_module, dst_module);
 		src_module = dst_module;
 	}
 
-	skl_delete_pipe(ctx, mconfig->pipe);
+	skl_delete_pipe(skl, mconfig->pipe);
 
 	list_for_each_entry(w_module, &s_pipe->w_list, node) {
 		src_module = w_module->w->priv;
 		src_module->m_state = SKL_MODULE_UNINIT;
 	}
 
-	return skl_tplg_unload_pipe_modules(ctx, s_pipe);
+	return skl_tplg_unload_pipe_modules(skl, s_pipe);
 }
 
 /*
@@ -1355,16 +1345,15 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
  *   - In source pipe is connected, unbind with source pipelines
  */
 static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
-								struct skl *skl)
+							struct skl_dev *skl)
 {
 	struct skl_module_cfg *src_mconfig, *sink_mconfig;
 	int ret = 0, i;
-	struct skl_sst *ctx = skl->skl_sst;
 
 	src_mconfig = w->priv;
 
 	/* Stop the pipe since this is a mixin module */
-	ret = skl_stop_pipe(ctx, src_mconfig->pipe);
+	ret = skl_stop_pipe(skl, src_mconfig->pipe);
 	if (ret)
 		return ret;
 
@@ -1377,7 +1366,7 @@ static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 			 * This is a connecter and if path is found that means
 			 * unbind between source and sink has not happened yet
 			 */
-			ret = skl_unbind_modules(ctx, src_mconfig,
+			ret = skl_unbind_modules(skl, src_mconfig,
 							sink_mconfig);
 		}
 	}
@@ -1395,7 +1384,7 @@ static int skl_tplg_mixer_event(struct snd_soc_dapm_widget *w,
 				struct snd_kcontrol *k, int event)
 {
 	struct snd_soc_dapm_context *dapm = w->dapm;
-	struct skl *skl = get_skl_ctx(dapm->dev);
+	struct skl_dev *skl = get_skl_ctx(dapm->dev);
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -1425,7 +1414,7 @@ static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w,
 
 {
 	struct snd_soc_dapm_context *dapm = w->dapm;
-	struct skl *skl = get_skl_ctx(dapm->dev);
+	struct skl_dev *skl = get_skl_ctx(dapm->dev);
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -1446,10 +1435,10 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol,
 	struct skl_algo_data *bc = (struct skl_algo_data *)sb->dobj.private;
 	struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol);
 	struct skl_module_cfg *mconfig = w->priv;
-	struct skl *skl = get_skl_ctx(w->dapm->dev);
+	struct skl_dev *skl = get_skl_ctx(w->dapm->dev);
 
 	if (w->power)
-		skl_get_module_params(skl->skl_sst, (u32 *)bc->params,
+		skl_get_module_params(skl, (u32 *)bc->params,
 				      bc->size, bc->param_id, mconfig);
 
 	/* decrement size for TLV header */
@@ -1481,7 +1470,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol,
 	struct soc_bytes_ext *sb =
 			(struct soc_bytes_ext *)kcontrol->private_value;
 	struct skl_algo_data *ac = (struct skl_algo_data *)sb->dobj.private;
-	struct skl *skl = get_skl_ctx(w->dapm->dev);
+	struct skl_dev *skl = get_skl_ctx(w->dapm->dev);
 
 	if (ac->params) {
 		/*
@@ -1498,7 +1487,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol,
 			return -EFAULT;
 
 		if (w->power)
-			return skl_set_module_params(skl->skl_sst,
+			return skl_set_module_params(skl,
 						(u32 *)ac->params, ac->size,
 						ac->param_id, mconfig);
 	}
@@ -1659,7 +1648,7 @@ int skl_tplg_update_pipe_params(struct device *dev,
 			struct skl_pipe_params *params)
 {
 	struct skl_module_res *res = &mconfig->module->resources[0];
-	struct skl *skl = get_skl_ctx(dev);
+	struct skl_dev *skl = get_skl_ctx(dev);
 	struct skl_module_fmt *format = NULL;
 	u8 cfg_idx = mconfig->pipe->cur_config_idx;
 
@@ -1856,7 +1845,7 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai,
 				struct skl_pipe_params *params)
 {
 	struct nhlt_specific_cfg *cfg;
-	struct skl *skl = get_skl_ctx(dai->dev);
+	struct skl_dev *skl = get_skl_ctx(dai->dev);
 	int link_type = skl_tplg_be_link_type(mconfig->dev_type);
 	u8 dev_type = skl_tplg_be_dev_type(mconfig->dev_type);
 
@@ -2070,7 +2059,7 @@ static int skl_tplg_fill_pipe_tkn(struct device *dev,
  * Return an existing pipe if the pipe already exists.
  */
 static int skl_tplg_add_pipe(struct device *dev,
-		struct skl_module_cfg *mconfig, struct skl *skl,
+		struct skl_module_cfg *mconfig, struct skl_dev *skl,
 		struct snd_soc_tplg_vendor_value_elem *tkn_elem)
 {
 	struct skl_pipeline *ppl;
@@ -2381,7 +2370,7 @@ static int skl_tplg_fill_res_tkn(struct device *dev,
  */
 static int skl_tplg_get_token(struct device *dev,
 		struct snd_soc_tplg_vendor_value_elem *tkn_elem,
-		struct skl *skl, struct skl_module_cfg *mconfig)
+		struct skl_dev *skl, struct skl_module_cfg *mconfig)
 {
 	int tkn_count = 0;
 	int ret;
@@ -2631,7 +2620,7 @@ static int skl_tplg_get_token(struct device *dev,
  * module private data
  */
 static int skl_tplg_get_tokens(struct device *dev,
-		char *pvt_data,	struct skl *skl,
+		char *pvt_data,	struct skl_dev *skl,
 		struct skl_module_cfg *mconfig, int block_size)
 {
 	struct snd_soc_tplg_vendor_array *array;
@@ -2727,8 +2716,8 @@ static int skl_tplg_get_desc_blocks(struct device *dev,
  * Otherwise we create a new instance and add into driver list
  */
 static int skl_tplg_add_pipe_v4(struct device *dev,
-				struct skl_module_cfg *mconfig, struct skl *skl,
-				struct skl_dfw_v4_pipe *dfw_pipe)
+			struct skl_module_cfg *mconfig, struct skl_dev *skl,
+			struct skl_dfw_v4_pipe *dfw_pipe)
 {
 	struct skl_pipeline *ppl;
 	struct skl_pipe *pipe;
@@ -2804,7 +2793,7 @@ static void skl_tplg_fill_fmt_v4(struct skl_module_pin_fmt *dst_fmt,
 }
 
 static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w,
-				    struct skl *skl, struct device *dev,
+				    struct skl_dev *skl, struct device *dev,
 				    struct skl_module_cfg *mconfig)
 {
 	struct skl_dfw_v4_module *dfw =
@@ -2886,7 +2875,7 @@ static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w,
  * for the type and size of the suceeding data block.
  */
 static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w,
-				struct skl *skl, struct device *dev,
+				struct skl_dev *skl, struct device *dev,
 				struct skl_module_cfg *mconfig)
 {
 	struct snd_soc_tplg_vendor_array *array;
@@ -2981,9 +2970,8 @@ static void skl_clear_pin_config(struct snd_soc_component *component,
 	}
 }
 
-void skl_cleanup_resources(struct skl *skl)
+void skl_cleanup_resources(struct skl_dev *skl)
 {
-	struct skl_sst *ctx = skl->skl_sst;
 	struct snd_soc_component *soc_component = skl->component;
 	struct snd_soc_dapm_widget *w;
 	struct snd_soc_card *card;
@@ -2999,11 +2987,11 @@ void skl_cleanup_resources(struct skl *skl)
 	skl->resource.mcps = 0;
 
 	list_for_each_entry(w, &card->widgets, list) {
-		if (is_skl_dsp_widget_type(w, ctx->dev) && w->priv != NULL)
+		if (is_skl_dsp_widget_type(w, skl->dev) && w->priv != NULL)
 			skl_clear_pin_config(soc_component, w);
 	}
 
-	skl_clear_module_cnt(ctx->dsp);
+	skl_clear_module_cnt(skl->dsp);
 }
 
 /*
@@ -3019,7 +3007,7 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index,
 {
 	int ret;
 	struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct skl_module_cfg *mconfig;
 
 	if (!tplg_w->priv.size)
@@ -3163,21 +3151,21 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt,
 
 static int skl_tplg_fill_str_mfest_tkn(struct device *dev,
 		struct snd_soc_tplg_vendor_string_elem *str_elem,
-		struct skl *skl)
+		struct skl_dev *skl)
 {
 	int tkn_count = 0;
 	static int ref_count;
 
 	switch (str_elem->token) {
 	case SKL_TKN_STR_LIB_NAME:
-		if (ref_count > skl->skl_sst->lib_count - 1) {
+		if (ref_count > skl->lib_count - 1) {
 			ref_count = 0;
 			return -EINVAL;
 		}
 
-		strncpy(skl->skl_sst->lib_info[ref_count].name,
+		strncpy(skl->lib_info[ref_count].name,
 			str_elem->string,
-			ARRAY_SIZE(skl->skl_sst->lib_info[ref_count].name));
+			ARRAY_SIZE(skl->lib_info[ref_count].name));
 		ref_count++;
 		break;
 
@@ -3192,7 +3180,7 @@ static int skl_tplg_fill_str_mfest_tkn(struct device *dev,
 
 static int skl_tplg_get_str_tkn(struct device *dev,
 		struct snd_soc_tplg_vendor_array *array,
-		struct skl *skl)
+		struct skl_dev *skl)
 {
 	int tkn_count = 0, ret;
 	struct snd_soc_tplg_vendor_string_elem *str_elem;
@@ -3299,7 +3287,7 @@ static int skl_tplg_fill_mod_info(struct device *dev,
 
 static int skl_tplg_get_int_tkn(struct device *dev,
 		struct snd_soc_tplg_vendor_value_elem *tkn_elem,
-		struct skl *skl)
+		struct skl_dev *skl)
 {
 	int tkn_count = 0, ret;
 	static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx;
@@ -3319,7 +3307,7 @@ static int skl_tplg_get_int_tkn(struct device *dev,
 
 	switch (tkn_elem->token) {
 	case SKL_TKN_U32_LIB_COUNT:
-		skl->skl_sst->lib_count = tkn_elem->value;
+		skl->lib_count = tkn_elem->value;
 		break;
 
 	case SKL_TKN_U8_NUM_MOD:
@@ -3466,7 +3454,7 @@ static int skl_tplg_get_int_tkn(struct device *dev,
 }
 
 static int skl_tplg_get_manifest_uuid(struct device *dev,
-				struct skl *skl,
+				struct skl_dev *skl,
 				struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn)
 {
 	static int ref_count;
@@ -3489,7 +3477,7 @@ static int skl_tplg_get_manifest_uuid(struct device *dev,
  * type.
  */
 static int skl_tplg_get_manifest_tkn(struct device *dev,
-		char *pvt_data, struct skl *skl,
+		char *pvt_data, struct skl_dev *skl,
 		int block_size)
 {
 	int tkn_count = 0, ret;
@@ -3550,7 +3538,7 @@ static int skl_tplg_get_manifest_tkn(struct device *dev,
  * preceded by descriptors for type and size of data block.
  */
 static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest,
-			struct device *dev, struct skl *skl)
+			struct device *dev, struct skl_dev *skl)
 {
 	struct snd_soc_tplg_vendor_array *array;
 	int num_blocks, block_size = 0, block_type, off = 0;
@@ -3612,7 +3600,7 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int index,
 				struct snd_soc_tplg_manifest *manifest)
 {
 	struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	/* proceed only if we have private data defined */
 	if (manifest->priv.size == 0)
@@ -3620,9 +3608,9 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int index,
 
 	skl_tplg_get_manifest_data(manifest, bus->dev, skl);
 
-	if (skl->skl_sst->lib_count > SKL_MAX_LIB) {
+	if (skl->lib_count > SKL_MAX_LIB) {
 		dev_err(bus->dev, "Exceeding max Library count. Got:%d\n",
-					skl->skl_sst->lib_count);
+					skl->lib_count);
 		return  -EINVAL;
 	}
 
@@ -3671,7 +3659,7 @@ static int skl_tplg_create_pipe_widget_list(struct snd_soc_component *component)
 	return 0;
 }
 
-static void skl_tplg_set_pipe_type(struct skl *skl, struct skl_pipe *pipe)
+static void skl_tplg_set_pipe_type(struct skl_dev *skl, struct skl_pipe *pipe)
 {
 	struct skl_pipe_module *w_module;
 	struct snd_soc_dapm_widget *w;
@@ -3705,7 +3693,7 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus)
 {
 	int ret;
 	const struct firmware *fw;
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct skl_pipeline *ppl;
 
 	ret = request_firmware(&fw, skl->tplg_name, bus->dev);
@@ -3748,7 +3736,7 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus)
 
 void skl_tplg_exit(struct snd_soc_component *component, struct hdac_bus *bus)
 {
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct skl_pipeline *ppl, *tmp;
 
 	if (!list_empty(&skl->ppl_list))
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 665e35cee50d..0a4d89cdaa4c 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -447,7 +447,7 @@ enum skl_channel {
 	SKL_CH_QUATRO = 4,
 };
 
-static inline struct skl *get_skl_ctx(struct device *dev)
+static inline struct skl_dev *get_skl_ctx(struct device *dev)
 {
 	struct hdac_bus *bus = dev_get_drvdata(dev);
 
@@ -456,7 +456,7 @@ static inline struct skl *get_skl_ctx(struct device *dev)
 
 int skl_tplg_be_update_params(struct snd_soc_dai *dai,
 	struct skl_pipe_params *params);
-int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps,
+int skl_dsp_set_dma_control(struct skl_dev *skl, u32 *caps,
 			u32 caps_size, u32 node_id);
 void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai,
 	struct skl_pipe_params *params, int stream);
@@ -469,32 +469,32 @@ struct skl_module_cfg *skl_tplg_fe_get_cpr_module(
 int skl_tplg_update_pipe_params(struct device *dev,
 		struct skl_module_cfg *mconfig, struct skl_pipe_params *params);
 
-void skl_tplg_d0i3_get(struct skl *skl, enum d0i3_capability caps);
-void skl_tplg_d0i3_put(struct skl *skl, enum d0i3_capability caps);
+void skl_tplg_d0i3_get(struct skl_dev *skl, enum d0i3_capability caps);
+void skl_tplg_d0i3_put(struct skl_dev *skl, enum d0i3_capability caps);
 
-int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe);
+int skl_create_pipeline(struct skl_dev *skl, struct skl_pipe *pipe);
 
-int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
+int skl_run_pipe(struct skl_dev *skl, struct skl_pipe *pipe);
 
-int skl_pause_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
+int skl_pause_pipe(struct skl_dev *skl, struct skl_pipe *pipe);
 
-int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
+int skl_delete_pipe(struct skl_dev *skl, struct skl_pipe *pipe);
 
-int skl_stop_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
+int skl_stop_pipe(struct skl_dev *skl, struct skl_pipe *pipe);
 
-int skl_reset_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
+int skl_reset_pipe(struct skl_dev *skl, struct skl_pipe *pipe);
 
-int skl_init_module(struct skl_sst *ctx, struct skl_module_cfg *module_config);
+int skl_init_module(struct skl_dev *skl, struct skl_module_cfg *module_config);
 
-int skl_bind_modules(struct skl_sst *ctx, struct skl_module_cfg
+int skl_bind_modules(struct skl_dev *skl, struct skl_module_cfg
 	*src_module, struct skl_module_cfg *dst_module);
 
-int skl_unbind_modules(struct skl_sst *ctx, struct skl_module_cfg
+int skl_unbind_modules(struct skl_dev *skl, struct skl_module_cfg
 	*src_module, struct skl_module_cfg *dst_module);
 
-int skl_set_module_params(struct skl_sst *ctx, u32 *params, int size,
+int skl_set_module_params(struct skl_dev *skl, u32 *params, int size,
 			u32 param_id, struct skl_module_cfg *mcfg);
-int skl_get_module_params(struct skl_sst *ctx, u32 *params, int size,
+int skl_get_module_params(struct skl_dev *skl, u32 *params, int size,
 			  u32 param_id, struct skl_module_cfg *mcfg);
 
 struct skl_module_cfg *skl_tplg_be_get_cpr_module(struct snd_soc_dai *dai,
@@ -508,6 +508,6 @@ int skl_pcm_link_dma_prepare(struct device *dev,
 int skl_dai_load(struct snd_soc_component *cmp, int index,
 		struct snd_soc_dai_driver *dai_drv,
 		struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai);
-void skl_tplg_add_moduleid_in_bind_params(struct skl *skl,
+void skl_tplg_add_moduleid_in_bind_params(struct skl_dev *skl,
 				struct snd_soc_dapm_widget *w);
 #endif
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 3362e71b4563..6be25e617c21 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -50,7 +50,7 @@ static void skl_update_pci_byte(struct pci_dev *pci, unsigned int reg,
 	pci_write_config_byte(pci, reg, data);
 }
 
-static void skl_init_pci(struct skl *skl)
+static void skl_init_pci(struct skl_dev *skl)
 {
 	struct hdac_bus *bus = skl_to_bus(skl);
 
@@ -252,7 +252,7 @@ static irqreturn_t skl_threaded_handler(int irq, void *dev_id)
 
 static int skl_acquire_irq(struct hdac_bus *bus, int do_disconnect)
 {
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	int ret;
 
 	ret = request_threaded_irq(skl->pci->irq, skl_interrupt,
@@ -276,7 +276,7 @@ static int skl_suspend_late(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct hdac_bus *bus = pci_get_drvdata(pci);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	return skl_suspend_late_dsp(skl);
 }
@@ -284,7 +284,7 @@ static int skl_suspend_late(struct device *dev)
 #ifdef CONFIG_PM
 static int _skl_suspend(struct hdac_bus *bus)
 {
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct pci_dev *pci = to_pci_dev(bus->dev);
 	int ret;
 
@@ -307,7 +307,7 @@ static int _skl_suspend(struct hdac_bus *bus)
 
 static int _skl_resume(struct hdac_bus *bus)
 {
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	skl_init_pci(skl);
 	skl_dum_set(bus);
@@ -325,7 +325,7 @@ static int skl_suspend(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct hdac_bus *bus = pci_get_drvdata(pci);
-	struct skl *skl  = bus_to_skl(bus);
+	struct skl_dev *skl  = bus_to_skl(bus);
 	int ret;
 
 	/*
@@ -345,7 +345,7 @@ static int skl_suspend(struct device *dev)
 		ret = _skl_suspend(bus);
 		if (ret < 0)
 			return ret;
-		skl->skl_sst->fw_loaded = false;
+		skl->fw_loaded = false;
 	}
 
 	return 0;
@@ -355,7 +355,7 @@ static int skl_resume(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct hdac_bus *bus = pci_get_drvdata(pci);
-	struct skl *skl  = bus_to_skl(bus);
+	struct skl_dev *skl  = bus_to_skl(bus);
 	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
@@ -430,7 +430,7 @@ static const struct dev_pm_ops skl_pm = {
  */
 static int skl_free(struct hdac_bus *bus)
 {
-	struct skl *skl  = bus_to_skl(bus);
+	struct skl_dev *skl  = bus_to_skl(bus);
 
 	skl->init_done = 0; /* to be sure */
 
@@ -475,7 +475,7 @@ static struct skl_ssp_clk skl_ssp_clks[] = {
 						{.name = "ssp5_sclkfs"},
 };
 
-static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl,
+static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl,
 					struct snd_soc_acpi_mach *machines)
 {
 	struct hdac_bus *bus = skl_to_bus(skl);
@@ -494,7 +494,7 @@ static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl,
 	return mach;
 }
 
-static int skl_find_machine(struct skl *skl, void *driver_data)
+static int skl_find_machine(struct skl_dev *skl, void *driver_data)
 {
 	struct hdac_bus *bus = skl_to_bus(skl);
 	struct snd_soc_acpi_mach *mach = driver_data;
@@ -522,7 +522,7 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 	return 0;
 }
 
-static int skl_machine_device_register(struct skl *skl)
+static int skl_machine_device_register(struct skl_dev *skl)
 {
 	struct snd_soc_acpi_mach *mach = skl->mach;
 	struct hdac_bus *bus = skl_to_bus(skl);
@@ -558,13 +558,13 @@ static int skl_machine_device_register(struct skl *skl)
 	return 0;
 }
 
-static void skl_machine_device_unregister(struct skl *skl)
+static void skl_machine_device_unregister(struct skl_dev *skl)
 {
 	if (skl->i2s_dev)
 		platform_device_unregister(skl->i2s_dev);
 }
 
-static int skl_dmic_device_register(struct skl *skl)
+static int skl_dmic_device_register(struct skl_dev *skl)
 {
 	struct hdac_bus *bus = skl_to_bus(skl);
 	struct platform_device *pdev;
@@ -588,7 +588,7 @@ static int skl_dmic_device_register(struct skl *skl)
 	return 0;
 }
 
-static void skl_dmic_device_unregister(struct skl *skl)
+static void skl_dmic_device_unregister(struct skl_dev *skl)
 {
 	if (skl->dmic_dev)
 		platform_device_unregister(skl->dmic_dev);
@@ -626,7 +626,7 @@ static void init_skl_xtal_rate(int pci_id)
 	}
 }
 
-static int skl_clock_device_register(struct skl *skl)
+static int skl_clock_device_register(struct skl_dev *skl)
 {
 	struct platform_device_info pdevinfo = {NULL};
 	struct skl_clk_pdata *clk_pdata;
@@ -656,7 +656,7 @@ static int skl_clock_device_register(struct skl *skl)
 	return PTR_ERR_OR_ZERO(skl->clk_dev);
 }
 
-static void skl_clock_device_unregister(struct skl *skl)
+static void skl_clock_device_unregister(struct skl_dev *skl)
 {
 	if (skl->clk_dev)
 		platform_device_unregister(skl->clk_dev);
@@ -692,7 +692,7 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 	unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) |
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	unsigned int res = -1;
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	struct hdac_hda_priv *hda_codec;
 	int err;
@@ -792,7 +792,7 @@ static int skl_i915_init(struct hdac_bus *bus)
 
 static void skl_probe_work(struct work_struct *work)
 {
-	struct skl *skl = container_of(work, struct skl, probe_work);
+	struct skl_dev *skl = container_of(work, struct skl_dev, probe_work);
 	struct hdac_bus *bus = skl_to_bus(skl);
 	struct hdac_ext_link *hlink = NULL;
 	int err;
@@ -855,10 +855,10 @@ out_err:
  */
 static int skl_create(struct pci_dev *pci,
 		      const struct hdac_io_ops *io_ops,
-		      struct skl **rskl)
+		      struct skl_dev **rskl)
 {
 	struct hdac_ext_bus_ops *ext_ops = NULL;
-	struct skl *skl;
+	struct skl_dev *skl;
 	struct hdac_bus *bus;
 	struct hda_bus *hbus;
 	int err;
@@ -902,7 +902,7 @@ static int skl_create(struct pci_dev *pci,
 
 static int skl_first_init(struct hdac_bus *bus)
 {
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 	struct pci_dev *pci = skl->pci;
 	int err;
 	unsigned short gcap;
@@ -978,7 +978,7 @@ static int skl_first_init(struct hdac_bus *bus)
 static int skl_probe(struct pci_dev *pci,
 		     const struct pci_device_id *pci_id)
 {
-	struct skl *skl;
+	struct skl_dev *skl;
 	struct hdac_bus *bus = NULL;
 	int err;
 
@@ -1071,8 +1071,8 @@ static int skl_probe(struct pci_dev *pci,
 		dev_dbg(bus->dev, "error failed to register dsp\n");
 		goto out_nhlt_free;
 	}
-	skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge;
-	skl->skl_sst->clock_power_gating = skl_clock_power_gating;
+	skl->enable_miscbdcge = skl_enable_miscbdcge;
+	skl->clock_power_gating = skl_clock_power_gating;
 
 	if (bus->mlcap)
 		snd_hdac_ext_bus_get_ml_capabilities(bus);
@@ -1107,7 +1107,7 @@ static void skl_shutdown(struct pci_dev *pci)
 	struct hdac_bus *bus = pci_get_drvdata(pci);
 	struct hdac_stream *s;
 	struct hdac_ext_stream *stream;
-	struct skl *skl;
+	struct skl_dev *skl;
 
 	if (!bus)
 		return;
@@ -1129,7 +1129,7 @@ static void skl_shutdown(struct pci_dev *pci)
 static void skl_remove(struct pci_dev *pci)
 {
 	struct hdac_bus *bus = pci_get_drvdata(pci);
-	struct skl *skl = bus_to_skl(bus);
+	struct skl_dev *skl = bus_to_skl(bus);
 
 	cancel_work_sync(&skl->probe_work);
 
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 6070666a6392..d3d84f4125d9 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -18,6 +18,7 @@
 #include <sound/soc.h>
 #include "skl-nhlt.h"
 #include "skl-ssp-clk.h"
+#include "skl-sst-ipc.h"
 
 #define SKL_SUSPEND_DELAY 2000
 
@@ -63,7 +64,7 @@ struct skl_fw_config {
 	struct skl_astate_config *astate_cfg;
 };
 
-struct skl {
+struct skl_dev {
 	struct hda_bus hbus;
 	struct pci_dev *pci;
 
@@ -75,7 +76,6 @@ struct skl {
 	struct snd_soc_dai_driver *dais;
 
 	struct nhlt_acpi_table *nhlt; /* nhlt ptr */
-	struct skl_sst *skl_sst; /* sst skl ctx */
 
 	struct skl_dsp_resource resource;
 	struct list_head ppl_list;
@@ -96,13 +96,59 @@ struct skl {
 	bool use_tplg_pcm;
 	struct skl_fw_config cfg;
 	struct snd_soc_acpi_mach *mach;
+
+	struct device *dev;
+	struct sst_dsp *dsp;
+
+	/* boot */
+	wait_queue_head_t boot_wait;
+	bool boot_complete;
+
+	/* module load */
+	wait_queue_head_t mod_load_wait;
+	bool mod_load_complete;
+	bool mod_load_status;
+
+	/* IPC messaging */
+	struct sst_generic_ipc ipc;
+
+	/* callback for miscbdge */
+	void (*enable_miscbdcge)(struct device *dev, bool enable);
+	/* Is CGCTL.MISCBDCGE disabled */
+	bool miscbdcg_disabled;
+
+	/* Populate module information */
+	struct list_head uuid_list;
+
+	/* Is firmware loaded */
+	bool fw_loaded;
+
+	/* first boot ? */
+	bool is_first_boot;
+
+	/* multi-core */
+	struct skl_dsp_cores cores;
+
+	/* library info */
+	struct skl_lib_info  lib_info[SKL_MAX_LIB];
+	int lib_count;
+
+	/* Callback to update D0i3C register */
+	void (*update_d0i3c)(struct device *dev, bool enable);
+
+	struct skl_d0i3_data d0i3;
+
+	const struct skl_dsp_ops *dsp_ops;
+
+	/* Callback to update dynamic clock and power gating registers */
+	void (*clock_power_gating)(struct device *dev, bool enable);
 };
 
 #define skl_to_bus(s)  (&(s)->hbus.core)
-#define bus_to_skl(bus) container_of(bus, struct skl, hbus.core)
+#define bus_to_skl(bus) container_of(bus, struct skl_dev, hbus.core)
 
 #define skl_to_hbus(s) (&(s)->hbus)
-#define hbus_to_skl(hbus) container_of((hbus), struct skl, (hbus))
+#define hbus_to_skl(hbus) container_of((hbus), struct skl_dev, (hbus))
 
 /* to pass dai dma data */
 struct skl_dma_params {
@@ -121,9 +167,9 @@ struct skl_dsp_ops {
 	int (*init)(struct device *dev, void __iomem *mmio_base,
 			int irq, const char *fw_name,
 			struct skl_dsp_loader_ops loader_ops,
-			struct skl_sst **skl_sst);
-	int (*init_fw)(struct device *dev, struct skl_sst *ctx);
-	void (*cleanup)(struct device *dev, struct skl_sst *ctx);
+			struct skl_dev **skl_sst);
+	int (*init_fw)(struct device *dev, struct skl_dev *skl);
+	void (*cleanup)(struct device *dev, struct skl_dev *skl);
 };
 
 int skl_platform_unregister(struct device *dev);
@@ -131,42 +177,42 @@ int skl_platform_register(struct device *dev);
 
 struct nhlt_acpi_table *skl_nhlt_init(struct device *dev);
 void skl_nhlt_free(struct nhlt_acpi_table *addr);
-struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance,
+struct nhlt_specific_cfg *skl_get_ep_blob(struct skl_dev *skl, u32 instance,
 					u8 link_type, u8 s_fmt, u8 no_ch,
 					u32 s_rate, u8 dirn, u8 dev_type);
 
-int skl_get_dmic_geo(struct skl *skl);
-int skl_nhlt_update_topology_bin(struct skl *skl);
-int skl_init_dsp(struct skl *skl);
-int skl_free_dsp(struct skl *skl);
-int skl_suspend_late_dsp(struct skl *skl);
-int skl_suspend_dsp(struct skl *skl);
-int skl_resume_dsp(struct skl *skl);
-void skl_cleanup_resources(struct skl *skl);
+int skl_get_dmic_geo(struct skl_dev *skl);
+int skl_nhlt_update_topology_bin(struct skl_dev *skl);
+int skl_init_dsp(struct skl_dev *skl);
+int skl_free_dsp(struct skl_dev *skl);
+int skl_suspend_late_dsp(struct skl_dev *skl);
+int skl_suspend_dsp(struct skl_dev *skl);
+int skl_resume_dsp(struct skl_dev *skl);
+void skl_cleanup_resources(struct skl_dev *skl);
 const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id);
 void skl_update_d0i3c(struct device *dev, bool enable);
-int skl_nhlt_create_sysfs(struct skl *skl);
-void skl_nhlt_remove_sysfs(struct skl *skl);
-void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks);
+int skl_nhlt_create_sysfs(struct skl_dev *skl);
+void skl_nhlt_remove_sysfs(struct skl_dev *skl);
+void skl_get_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks);
 struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id);
-int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps,
+int skl_dsp_set_dma_control(struct skl_dev *skl, u32 *caps,
 				u32 caps_size, u32 node_id);
 
 struct skl_module_cfg;
 
 #ifdef CONFIG_DEBUG_FS
-struct skl_debug *skl_debugfs_init(struct skl *skl);
-void skl_debugfs_exit(struct skl *skl);
+struct skl_debug *skl_debugfs_init(struct skl_dev *skl);
+void skl_debugfs_exit(struct skl_dev *skl);
 void skl_debug_init_module(struct skl_debug *d,
 			struct snd_soc_dapm_widget *w,
 			struct skl_module_cfg *mconfig);
 #else
-static inline struct skl_debug *skl_debugfs_init(struct skl *skl)
+static inline struct skl_debug *skl_debugfs_init(struct skl_dev *skl)
 {
 	return NULL;
 }
 
-static inline void skl_debugfs_exit(struct skl *skl)
+static inline void skl_debugfs_exit(struct skl_dev *skl)
 {}
 
 static inline void skl_debug_init_module(struct skl_debug *d,
-- 
cgit v1.2.3-59-g8ed1b


From 88abcc90cb04cbabcc3e384e8d3431a49f56a2b3 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Date: Tue, 23 Jul 2019 16:58:49 +0200
Subject: ASoC: Intel: Skylake: Combine snd_soc_skl_ipc and snd_soc_skl
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

As both modules are core part of Skylake driver and none can live
without the other, combine snd_soc_skl_ipc and snd_soc_skl.

It's highly probable IPC module was to be treated as an interface for
platform specific code implementations e.g.: possibility of existence of
BXT specific code without SKL one. However, most funtionalities are
being inherited from one DSP firmware to another, and thus this
assumption fails.

skl-sst, bxt-sst and cnl-sst are not individuals pointing respectively
to SKL (cAVS 1.5), BXT (cAVS 1.5+) & CNL (cAVS 1.8) standalone
implementations. Code found within these is shared among all platforms
whenever necessary to avoid code duplication and reduce development
burden.

Merge also helps in cleaning up internal code in future changes.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723145854.8527-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/Makefile | 12 +++---------
 1 file changed, 3 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index 86f6e1d801af..48544ff1a3e6 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -1,6 +1,7 @@
 # SPDX-License-Identifier: GPL-2.0
-snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o \
-skl-topology.o
+snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o skl-topology.o \
+		skl-sst-ipc.o skl-sst-dsp.o cnl-sst-dsp.o skl-sst-cldma.o \
+		skl-sst.o bxt-sst.o cnl-sst.o skl-sst-utils.o
 
 ifdef CONFIG_DEBUG_FS
   snd-soc-skl-objs += skl-debug.o
@@ -8,13 +9,6 @@ endif
 
 obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
 
-# Skylake IPC Support
-snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o cnl-sst-dsp.o \
-		skl-sst-cldma.o skl-sst.o bxt-sst.o cnl-sst.o \
-		skl-sst-utils.o
-
-obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o
-
 #Skylake Clock device support
 snd-soc-skl-ssp-clk-objs := skl-ssp-clk.o
 
-- 
cgit v1.2.3-59-g8ed1b


From f2a167ca681d21aa46f127032cc005879c328823 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 23 Jul 2019 16:58:50 +0200
Subject: ASoC: Intel: Skylake: Remove MCPS available check

The entire logic for MCPS calculation and DSP scheduling is found
within DSP firmware. Currently driver implements simplistic, inaccurate
logic itself which may prevent pipeline creation despite firmware being
completely fine its parameters.

Remove that logic and leave the MCPS calculation to DSP alone.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723145854.8527-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 67 +---------------------------------
 sound/soc/intel/skylake/skl.h          |  2 -
 2 files changed, 1 insertion(+), 68 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index abf860109607..2af311150543 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -142,54 +142,6 @@ static void skl_tplg_alloc_pipe_mem(struct skl_dev *skl,
 	skl->resource.mem += mconfig->pipe->memory_pages;
 }
 
-/*
- * Pipeline needs needs DSP CPU resources for computation, this is
- * quantified in MCPS (Million Clocks Per Second) required for module/pipe
- *
- * Each pipelines needs mcps to be allocated. Check if we have mcps for this
- * pipe.
- */
-
-static bool skl_is_pipe_mcps_avail(struct skl_dev *skl,
-				struct skl_module_cfg *mconfig)
-{
-	u8 res_idx = mconfig->res_idx;
-	struct skl_module_res *res = &mconfig->module->resources[res_idx];
-
-	if (skl->resource.mcps + res->cps > skl->resource.max_mcps) {
-		dev_err(skl->dev,
-			"%s: module_id %d instance %d\n", __func__,
-			mconfig->id.module_id, mconfig->id.instance_id);
-		dev_err(skl->dev,
-			"exceeds ppl mcps available %d > mem %d\n",
-			skl->resource.max_mcps, skl->resource.mcps);
-		return false;
-	} else {
-		return true;
-	}
-}
-
-static void skl_tplg_alloc_pipe_mcps(struct skl_dev *skl,
-				struct skl_module_cfg *mconfig)
-{
-	u8 res_idx = mconfig->res_idx;
-	struct skl_module_res *res = &mconfig->module->resources[res_idx];
-
-	skl->resource.mcps += res->cps;
-}
-
-/*
- * Free the mcps when tearing down
- */
-static void
-skl_tplg_free_pipe_mcps(struct skl_dev *skl, struct skl_module_cfg *mconfig)
-{
-	u8 res_idx = mconfig->res_idx;
-	struct skl_module_res *res = &mconfig->module->resources[res_idx];
-
-	skl->resource.mcps -= res->cps;
-}
-
 /*
  * Free the memory when tearing down
  */
@@ -582,10 +534,6 @@ skl_tplg_init_pipe_modules(struct skl_dev *skl, struct skl_pipe *pipe)
 		mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx;
 		mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx;
 
-		/* check resource available */
-		if (!skl_is_pipe_mcps_avail(skl, mconfig))
-			return -ENOMEM;
-
 		if (mconfig->module->loadable && skl->dsp->fw_ops.load_mod) {
 			ret = skl->dsp->fw_ops.load_mod(skl->dsp,
 				mconfig->id.module_id, mconfig->guid);
@@ -627,7 +575,7 @@ skl_tplg_init_pipe_modules(struct skl_dev *skl, struct skl_pipe *pipe)
 			skl_put_pvt_id(skl, uuid_mod, &mconfig->id.pvt_id);
 			goto err;
 		}
-		skl_tplg_alloc_pipe_mcps(skl, mconfig);
+
 		ret = skl_tplg_set_module_params(w, skl);
 		if (ret < 0)
 			goto err;
@@ -752,10 +700,6 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	if (ret < 0)
 		return ret;
 
-	/* check resource available */
-	if (!skl_is_pipe_mcps_avail(skl, mconfig))
-		return -EBUSY;
-
 	if (!skl_is_pipe_mem_avail(skl, mconfig))
 		return -ENOMEM;
 
@@ -768,7 +712,6 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 		return ret;
 
 	skl_tplg_alloc_pipe_mem(skl, mconfig);
-	skl_tplg_alloc_pipe_mcps(skl, mconfig);
 
 	/* Init all pipe modules from source to sink */
 	ret = skl_tplg_init_pipe_modules(skl, s_pipe);
@@ -1264,7 +1207,6 @@ static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
 
 /*
  * in the Post-PMD event of mixer we need to do following:
- *   - Free the mcps used
  *   - Free the mem used
  *   - Unbind the modules within the pipeline
  *   - Delete the pipeline (modules are not required to be explicitly
@@ -1282,7 +1224,6 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 	if (s_pipe->state == SKL_PIPE_INVALID)
 		return -EINVAL;
 
-	skl_tplg_free_pipe_mcps(skl, mconfig);
 	skl_tplg_free_pipe_mem(skl, mconfig);
 
 	list_for_each_entry(w_module, &s_pipe->w_list, node) {
@@ -1317,8 +1258,6 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 	list_for_each_entry(w_module, &s_pipe->w_list, node) {
 		dst_module = w_module->w->priv;
 
-		if (mconfig->m_state >= SKL_MODULE_INIT_DONE)
-			skl_tplg_free_pipe_mcps(skl, dst_module);
 		if (src_module == NULL) {
 			src_module = dst_module;
 			continue;
@@ -1340,7 +1279,6 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 
 /*
  * in the Post-PMD event of PGA we need to do following:
- *   - Free the mcps used
  *   - Stop the pipeline
  *   - In source pipe is connected, unbind with source pipelines
  */
@@ -2984,7 +2922,6 @@ void skl_cleanup_resources(struct skl_dev *skl)
 		return;
 
 	skl->resource.mem = 0;
-	skl->resource.mcps = 0;
 
 	list_for_each_entry(w, &card->widgets, list) {
 		if (is_skl_dsp_widget_type(w, skl->dev) && w->priv != NULL)
@@ -3683,7 +3620,6 @@ static void skl_tplg_set_pipe_type(struct skl_dev *skl, struct skl_pipe *pipe)
 }
 
 /* This will be read from topology manifest, currently defined here */
-#define SKL_MAX_MCPS 30000000
 #define SKL_FW_MAX_MEM 1000000
 
 /*
@@ -3720,7 +3656,6 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus)
 		return -EINVAL;
 	}
 
-	skl->resource.max_mcps = SKL_MAX_MCPS;
 	skl->resource.max_mem = SKL_FW_MAX_MEM;
 
 	skl->tplg = fw;
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index d3d84f4125d9..21c33c1af934 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -42,9 +42,7 @@
 #define AZX_REG_VS_EM2_L1SEN		BIT(13)
 
 struct skl_dsp_resource {
-	u32 max_mcps;
 	u32 max_mem;
-	u32 mcps;
 	u32 mem;
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From e5c29dc5221c328b6e54c5b2aba1feca4003c051 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 23 Jul 2019 16:58:51 +0200
Subject: ASoC: Intel: Skylake: Remove memory available check

Current memory availability check is a stub, while actual memory
management takes place in firmware. Leave this task to firmware entirely
and remove redundant code.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723145854.8527-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 59 ----------------------------------
 sound/soc/intel/skylake/skl.h          |  6 ----
 2 files changed, 65 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 2af311150543..53a024c0464d 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -109,49 +109,6 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w,
 	}
 }
 
-/*
- * Each pipelines needs memory to be allocated. Check if we have free memory
- * from available pool.
- */
-static bool skl_is_pipe_mem_avail(struct skl_dev *skl,
-				struct skl_module_cfg *mconfig)
-{
-	if (skl->resource.mem + mconfig->pipe->memory_pages >
-				skl->resource.max_mem) {
-		dev_err(skl->dev,
-				"%s: module_id %d instance %d\n", __func__,
-				mconfig->id.module_id,
-				mconfig->id.instance_id);
-		dev_err(skl->dev,
-				"exceeds ppl memory available %d mem %d\n",
-				skl->resource.max_mem, skl->resource.mem);
-		return false;
-	} else {
-		return true;
-	}
-}
-
-/*
- * Add the mem to the mem pool. This is freed when pipe is deleted.
- * Note: DSP does actual memory management we only keep track for complete
- * pool
- */
-static void skl_tplg_alloc_pipe_mem(struct skl_dev *skl,
-				struct skl_module_cfg *mconfig)
-{
-	skl->resource.mem += mconfig->pipe->memory_pages;
-}
-
-/*
- * Free the memory when tearing down
- */
-static void
-skl_tplg_free_pipe_mem(struct skl_dev *skl, struct skl_module_cfg *mconfig)
-{
-	skl->resource.mem -= mconfig->pipe->memory_pages;
-}
-
-
 static void skl_dump_mconfig(struct skl_dev *skl, struct skl_module_cfg *mcfg)
 {
 	struct skl_module_iface *iface = &mcfg->module->formats[0];
@@ -681,7 +638,6 @@ skl_tplg_get_pipe_config(struct skl_dev *skl, struct skl_module_cfg *mconfig)
 /*
  * Mixer module represents a pipeline. So in the Pre-PMU event of mixer we
  * need create the pipeline. So we do following:
- *   - check the resources
  *   - Create the pipeline
  *   - Initialize the modules in pipeline
  *   - finally bind all modules together
@@ -700,9 +656,6 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	if (ret < 0)
 		return ret;
 
-	if (!skl_is_pipe_mem_avail(skl, mconfig))
-		return -ENOMEM;
-
 	/*
 	 * Create a list of modules for pipe.
 	 * This list contains modules from source to sink
@@ -711,8 +664,6 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	if (ret < 0)
 		return ret;
 
-	skl_tplg_alloc_pipe_mem(skl, mconfig);
-
 	/* Init all pipe modules from source to sink */
 	ret = skl_tplg_init_pipe_modules(skl, s_pipe);
 	if (ret < 0)
@@ -1207,7 +1158,6 @@ static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
 
 /*
  * in the Post-PMD event of mixer we need to do following:
- *   - Free the mem used
  *   - Unbind the modules within the pipeline
  *   - Delete the pipeline (modules are not required to be explicitly
  *     deleted, pipeline delete is enough here
@@ -1224,8 +1174,6 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 	if (s_pipe->state == SKL_PIPE_INVALID)
 		return -EINVAL;
 
-	skl_tplg_free_pipe_mem(skl, mconfig);
-
 	list_for_each_entry(w_module, &s_pipe->w_list, node) {
 		if (list_empty(&skl->bind_list))
 			break;
@@ -2921,8 +2869,6 @@ void skl_cleanup_resources(struct skl_dev *skl)
 	if (!card || !card->instantiated)
 		return;
 
-	skl->resource.mem = 0;
-
 	list_for_each_entry(w, &card->widgets, list) {
 		if (is_skl_dsp_widget_type(w, skl->dev) && w->priv != NULL)
 			skl_clear_pin_config(soc_component, w);
@@ -3619,9 +3565,6 @@ static void skl_tplg_set_pipe_type(struct skl_dev *skl, struct skl_pipe *pipe)
 		pipe->passthru = false;
 }
 
-/* This will be read from topology manifest, currently defined here */
-#define SKL_FW_MAX_MEM 1000000
-
 /*
  * SKL topology init routine
  */
@@ -3656,8 +3599,6 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus)
 		return -EINVAL;
 	}
 
-	skl->resource.max_mem = SKL_FW_MAX_MEM;
-
 	skl->tplg = fw;
 	ret = skl_tplg_create_pipe_widget_list(component);
 	if (ret < 0)
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 21c33c1af934..600a61f79b0a 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -41,11 +41,6 @@
 #define AZX_VS_EM2_DUM			BIT(23)
 #define AZX_REG_VS_EM2_L1SEN		BIT(13)
 
-struct skl_dsp_resource {
-	u32 max_mem;
-	u32 mem;
-};
-
 struct skl_debug;
 
 struct skl_astate_param {
@@ -75,7 +70,6 @@ struct skl_dev {
 
 	struct nhlt_acpi_table *nhlt; /* nhlt ptr */
 
-	struct skl_dsp_resource resource;
 	struct list_head ppl_list;
 	struct list_head bind_list;
 
-- 
cgit v1.2.3-59-g8ed1b


From d4e23b7c87e3439f6d0ec0c65ad7f919d6f4c124 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Date: Tue, 23 Jul 2019 16:58:52 +0200
Subject: ASoC: Intel: Skylake: Do not disable FW notifications
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

As per FW team recommendation we should not disable notifications.
By default, all notifications are enabled in DSP firmware. These
notifications provide a vital information whenever an error occurs.

Currently, driver disables them during boot sequences. By doing so,
Skylake may silently ignore severe stream errors.
Correct that by removing permissive code.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723145854.8527-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 20 --------------------
 sound/soc/intel/skylake/skl-pcm.c      |  1 -
 sound/soc/intel/skylake/skl-sst-dsp.h  |  1 -
 sound/soc/intel/skylake/skl-topology.h |  5 -----
 4 files changed, 27 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index f604d9c6824d..07762543fb13 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -58,24 +58,6 @@ void skl_dsp_set_astate_cfg(struct skl_dev *skl, u32 cnt, void *data)
 	skl_ipc_set_large_config(&skl->ipc, &msg, data);
 }
 
-#define NOTIFICATION_PARAM_ID 3
-#define NOTIFICATION_MASK 0xf
-
-/* disable notfication for underruns/overruns from firmware module */
-void skl_dsp_enable_notification(struct skl_dev *skl, bool enable)
-{
-	struct notification_mask mask;
-	struct skl_ipc_large_config_msg	msg = {0};
-
-	mask.notify = NOTIFICATION_MASK;
-	mask.enable = enable;
-
-	msg.large_param_id = NOTIFICATION_PARAM_ID;
-	msg.param_data_size = sizeof(mask);
-
-	skl_ipc_set_large_config(&skl->ipc, &msg, (u32 *)&mask);
-}
-
 static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
 				int stream_tag, int enable)
 {
@@ -436,8 +418,6 @@ int skl_resume_dsp(struct skl_dev *skl)
 	if (ret < 0)
 		return ret;
 
-	skl_dsp_enable_notification(skl, false);
-
 	if (skl->cfg.astate_cfg != NULL) {
 		skl_dsp_set_astate_cfg(skl, skl->cfg.astate_cfg->count,
 					skl->cfg.astate_cfg);
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 2ae5d09d7228..7f287424af9b 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -1445,7 +1445,6 @@ static int skl_platform_soc_probe(struct snd_soc_component *component)
 		}
 		skl_populate_modules(skl);
 		skl->update_d0i3c = skl_update_d0i3c;
-		skl_dsp_enable_notification(skl, false);
 
 		if (skl->cfg.astate_cfg != NULL) {
 			skl_dsp_set_astate_cfg(skl,
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index ed968f1042c7..cdfec0fca577 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -240,7 +240,6 @@ int skl_get_pvt_instance_id_map(struct skl_dev *skl,
 void skl_freeup_uuid_list(struct skl_dev *skl);
 
 int skl_dsp_strip_extended_manifest(struct firmware *fw);
-void skl_dsp_enable_notification(struct skl_dev *skl, bool enable);
 
 void skl_dsp_set_astate_cfg(struct skl_dev *skl, u32 cnt, void *data);
 
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 0a4d89cdaa4c..e2a2fc5c5545 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -140,11 +140,6 @@ struct skl_src_module_cfg {
 	enum skl_s_freq src_cfg;
 } __packed;
 
-struct notification_mask {
-	u32 notify;
-	u32 enable;
-} __packed;
-
 struct skl_up_down_mixer_cfg {
 	struct skl_base_cfg base_cfg;
 	enum skl_ch_cfg out_ch_cfg;
-- 
cgit v1.2.3-59-g8ed1b


From 84b71067ea840fadee32588aa3967d0d8c4e0b9a Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 23 Jul 2019 16:58:53 +0200
Subject: ASoC: Intel: Skylake: Make MCPS and CPS params obsolete

As per FW Interface Modules Configuration, init instance IPC request
requires base initial module configuration. This configuration structure
is made of:
- cpc (chunks per cycle)
- ibs (input buffer size)
- obs (output buffer size)
- is_pages (memory pages required)
- audio_fmt (self explanatory)

Skylake topology accepts following tokens: MCPS, CPS and CPC. All of
these are directly connected. Moreover, assigning one of these allows
to calculate the remaining two. In simplest scenario and assuming 1ms
scheduling, following is true:

CPS = CPC times 1000
MCPS = CPS times 1000 000
Note: these calculations vary depending on scenario and scheduling
requirements.

Given the current implementation, userspace is allowed to provide
different values for all three causing informational chaos. On top of
that, struct skl_base_cfg which represents base module configuration,
incorrectly takes CPS param instead of CPC.

This ambiguity may lead to user unintentionally providing improper
values to DSP firmware and thus impacting module scheduling in
unexpected fashion. Fix by making MCPS and CPS topology params obsolete
and relying solely on CPC value.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723145854.8527-7-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-debug.c    |  6 ++++--
 sound/soc/intel/skylake/skl-messages.c |  2 +-
 sound/soc/intel/skylake/skl-topology.c | 15 ++++++---------
 sound/soc/intel/skylake/skl-topology.h |  4 +---
 4 files changed, 12 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index c43aa4081232..fb232428109f 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -66,6 +66,8 @@ static ssize_t module_read(struct file *file, char __user *user_buf,
 			   size_t count, loff_t *ppos)
 {
 	struct skl_module_cfg *mconfig = file->private_data;
+	struct skl_module *module = mconfig->module;
+	struct skl_module_res *res = &module->resources[mconfig->res_idx];
 	char *buf;
 	ssize_t ret;
 
@@ -79,8 +81,8 @@ static ssize_t module_read(struct file *file, char __user *user_buf,
 			mconfig->id.pvt_id);
 
 	ret += snprintf(buf + ret, MOD_BUF - ret,
-			"Resources:\n\tMCPS %#x\n\tIBS %#x\n\tOBS %#x\t\n",
-			mconfig->mcps, mconfig->ibs, mconfig->obs);
+			"Resources:\n\tCPC %#x\n\tIBS %#x\n\tOBS %#x\t\n",
+			res->cpc, res->ibs, res->obs);
 
 	ret += snprintf(buf + ret, MOD_BUF - ret,
 			"Module data:\n\tCore %d\n\tIn queue %d\n\t"
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 07762543fb13..e8cc710f092b 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -477,7 +477,7 @@ static void skl_set_base_module_format(struct skl_dev *skl,
 
 	base_cfg->audio_fmt.interleaving = format->interleaving_style;
 
-	base_cfg->cps = res->cps;
+	base_cfg->cpc = res->cpc;
 	base_cfg->ibs = res->ibs;
 	base_cfg->obs = res->obs;
 	base_cfg->is_pages = res->is_pages;
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 53a024c0464d..118866cd5075 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -2205,10 +2205,6 @@ static int skl_tplg_fill_res_tkn(struct device *dev,
 		return -EINVAL;
 
 	switch (tkn_elem->token) {
-	case SKL_TKN_MM_U32_CPS:
-		res->cps = tkn_elem->value;
-		break;
-
 	case SKL_TKN_MM_U32_DMA_SIZE:
 		res->dma_buffer_size = tkn_elem->value;
 		break;
@@ -2229,10 +2225,6 @@ static int skl_tplg_fill_res_tkn(struct device *dev,
 		res->ibs = tkn_elem->value;
 		break;
 
-	case SKL_TKN_U32_MAX_MCPS:
-		res->cps = tkn_elem->value;
-		break;
-
 	case SKL_TKN_MM_U32_RES_PIN_ID:
 	case SKL_TKN_MM_U32_PIN_BUF:
 		ret = skl_tplg_manifest_pin_res_tkn(dev, tkn_elem, res,
@@ -2241,6 +2233,11 @@ static int skl_tplg_fill_res_tkn(struct device *dev,
 			return ret;
 		break;
 
+	case SKL_TKN_MM_U32_CPS:
+	case SKL_TKN_U32_MAX_MCPS:
+		/* ignore unused tokens */
+		break;
+
 	default:
 		dev_err(dev, "Not a res type token: %d", tkn_elem->token);
 		return -EINVAL;
@@ -2693,7 +2690,7 @@ static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w,
 		return ret;
 	mconfig->id.module_id = -1;
 	mconfig->id.instance_id = dfw->instance_id;
-	mconfig->module->resources[0].cps = dfw->max_mcps;
+	mconfig->module->resources[0].cpc = dfw->max_mcps / 1000;
 	mconfig->module->resources[0].ibs = dfw->ibs;
 	mconfig->module->resources[0].obs = dfw->obs;
 	mconfig->core_id = dfw->core_id;
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index e2a2fc5c5545..99a0277191ca 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -101,7 +101,7 @@ struct skl_audio_data_format {
 } __packed;
 
 struct skl_base_cfg {
-	u32 cps;
+	u32 cpc;
 	u32 ibs;
 	u32 obs;
 	u32 is_pages;
@@ -343,7 +343,6 @@ struct skl_module_pin_resources {
 struct skl_module_res {
 	u8 id;
 	u32 is_pages;
-	u32 cps;
 	u32 ibs;
 	u32 obs;
 	u32 dma_buffer_size;
@@ -384,7 +383,6 @@ struct skl_module_cfg {
 	u8 out_queue_mask;
 	u8 in_queue;
 	u8 out_queue;
-	u32 mcps;
 	u32 ibs;
 	u32 obs;
 	u8 is_loadable;
-- 
cgit v1.2.3-59-g8ed1b


From bdc873feec900005d52f9afd3a63290531c0c1f4 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 23 Jul 2019 16:58:54 +0200
Subject: ASoC: Intel: Skylake: Cleanup skl_module_cfg declaration

With removal of MCPS, CPS and CPC ambiguity, ibs and obs params for
struct skl_module_cfg have been left unused. Update struct declaration
by removing these two.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723145854.8527-8-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.h | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 99a0277191ca..e967800dbb62 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -383,8 +383,6 @@ struct skl_module_cfg {
 	u8 out_queue_mask;
 	u8 in_queue;
 	u8 out_queue;
-	u32 ibs;
-	u32 obs;
 	u8 is_loadable;
 	u8 core_id;
 	u8 dev_type;
-- 
cgit v1.2.3-59-g8ed1b


From f2c8b57014038ae064a0f918f939af510f0aed02 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 25 Jul 2019 11:33:21 +0300
Subject: ASoC: pcm3168a: Allow all channels in case of parallel DIN/DOUT setup

If multi DIN/DOUT mode is selected (tdm_slots == 2) then configure the
channel constraint to allow all channels.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083321.6776-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/pcm3168a.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 5d59ce254821..e84a1509fe65 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -599,6 +599,10 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream,
 				     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
 				     sample_min, 32);
 
+	/* Allow all channels in multi DIN/DOUT mode */
+	if (pcm3168a->tdm_slots == 2)
+		channel_max = channel_maxs[tx];
+
 	snd_pcm_hw_constraint_minmax(substream->runtime,
 				     SNDRV_PCM_HW_PARAM_CHANNELS,
 				     2, channel_max);
-- 
cgit v1.2.3-59-g8ed1b


From 816fe206138a7e3895c3a74fbd326a51df003f70 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 25 Jul 2019 11:34:11 +0300
Subject: ASoC: ti: davinci-mcasp: Add support for RIGHT_J format

The formater unit's rotation needs to be programmed differently for right
aligned bus format to have the data moved to the correct place.

Take the opportunity and simplify the formater unit setup code.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083411.7211-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-mcasp.c | 43 +++++++++++++++++++------------------------
 1 file changed, 19 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index ac59b509ead5..09ba3d3c70a0 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -465,6 +465,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* FS need to be inverted */
 		inv_fs = true;
 		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
 	case SND_SOC_DAIFMT_LEFT_J:
 		/* configure a full-word SYNC pulse (LRCLK) */
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
@@ -758,34 +759,28 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
 				       int sample_width)
 {
 	u32 fmt;
-	u32 tx_rotate = (sample_width / 4) & 0x7;
+	u32 tx_rotate, rx_rotate, slot_width;
 	u32 mask = (1ULL << sample_width) - 1;
-	u32 slot_width = sample_width;
-
-	/*
-	 * For captured data we should not rotate, inversion and masking is
-	 * enoguh to get the data to the right position:
-	 * Format	  data from bus		after reverse (XRBUF)
-	 * S16_LE:	|LSB|MSB|xxx|xxx|	|xxx|xxx|MSB|LSB|
-	 * S24_3LE:	|LSB|DAT|MSB|xxx|	|xxx|MSB|DAT|LSB|
-	 * S24_LE:	|LSB|DAT|MSB|xxx|	|xxx|MSB|DAT|LSB|
-	 * S32_LE:	|LSB|DAT|DAT|MSB|	|MSB|DAT|DAT|LSB|
-	 */
-	u32 rx_rotate = 0;
 
+	if (mcasp->slot_width)
+		slot_width = mcasp->slot_width;
+	else
+		slot_width = sample_width;
 	/*
-	 * Setting the tdm slot width either with set_clkdiv() or
-	 * set_tdm_slot() allows us to for example send 32 bits per
-	 * channel to the codec, while only 16 of them carry audio
-	 * payload.
+	 * TX rotation:
+	 * right aligned formats: rotate w/ slot_width
+	 * left aligned formats: rotate w/ sample_width
+	 *
+	 * RX rotation:
+	 * right aligned formats: no rotation needed
+	 * left aligned formats: rotate w/ (slot_width - sample_width)
 	 */
-	if (mcasp->slot_width) {
-		/*
-		 * When we have more bclk then it is needed for the
-		 * data, we need to use the rotation to move the
-		 * received samples to have correct alignment.
-		 */
-		slot_width = mcasp->slot_width;
+	if ((mcasp->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+	    SND_SOC_DAIFMT_RIGHT_J) {
+		tx_rotate = (slot_width / 4) & 0x7;
+		rx_rotate = 0;
+	} else {
+		tx_rotate = (sample_width / 4) & 0x7;
 		rx_rotate = (slot_width - sample_width) / 4;
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From b7989e271e9f7fd4edb4166cd2a79b980c4b6136 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 25 Jul 2019 11:34:32 +0300
Subject: ASoC: ti: davinci-mcasp: Improve serializer handling in multi AXR
 setups

When multiple serializers are used we need to track the number of
serializers used by the other stream direction to avoid killing data lines
when the first stream used more serializers than the second would need.
We are still protected against the case when the second stream uses more
serializers which had affected the running stream as well.

To take advantage of the improved serializer logic we need to modify the
channel constraints rule as well to allow the use of multiple serializers
for the second stream as additional ones will not affect the FS/BCLK on
the bus.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083432.7419-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-mcasp.c | 92 +++++++++++++++++++++++++-------------------
 1 file changed, 53 insertions(+), 39 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 09ba3d3c70a0..c3a2c53fdb3a 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -109,6 +109,7 @@ struct davinci_mcasp {
 
 	/* Used for comstraint setting on the second stream */
 	u32	channels;
+	u8	active_serializers[2];
 
 #ifdef CONFIG_GPIOLIB
 	struct gpio_chip gpio_chip;
@@ -813,6 +814,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	u8 rx_ser = 0;
 	u8 slots = mcasp->tdm_slots;
 	u8 max_active_serializers = (channels + slots - 1) / slots;
+	u8 max_rx_serializers, max_tx_serializers;
 	int active_serializers, numevt;
 	u32 reg;
 	/* Default configuration */
@@ -822,22 +824,28 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
+		max_tx_serializers = max_active_serializers;
+		max_rx_serializers =
+			mcasp->active_serializers[SNDRV_PCM_STREAM_CAPTURE];
 	} else {
 		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(mcasp, DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS);
+		max_tx_serializers =
+			mcasp->active_serializers[SNDRV_PCM_STREAM_PLAYBACK];
+		max_rx_serializers = max_active_serializers;
 	}
 
 	for (i = 0; i < mcasp->num_serializer; i++) {
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
 			       mcasp->serial_dir[i]);
 		if (mcasp->serial_dir[i] == TX_MODE &&
-					tx_ser < max_active_serializers) {
+					tx_ser < max_tx_serializers) {
 			mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
 				       mcasp->dismod, DISMOD_MASK);
 			set_bit(PIN_BIT_AXR(i), &mcasp->pdir);
 			tx_ser++;
 		} else if (mcasp->serial_dir[i] == RX_MODE &&
-					rx_ser < max_active_serializers) {
+					rx_ser < max_rx_serializers) {
 			clear_bit(PIN_BIT_AXR(i), &mcasp->pdir);
 			rx_ser++;
 		} else {
@@ -884,7 +892,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 		} else {
 			dma_data->maxburst = 0;
 		}
-		return 0;
+
+		goto out;
 	}
 
 	if (period_words % active_serializers) {
@@ -914,6 +923,9 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 		numevt = 0;
 	dma_data->maxburst = numevt;
 
+out:
+	mcasp->active_serializers[stream] = active_serializers;
+
 	return 0;
 }
 
@@ -1153,6 +1165,37 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	int period_size = params_period_size(params);
 	int ret;
 
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U8:
+	case SNDRV_PCM_FORMAT_S8:
+		word_length = 8;
+		break;
+
+	case SNDRV_PCM_FORMAT_U16_LE:
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word_length = 16;
+		break;
+
+	case SNDRV_PCM_FORMAT_U24_3LE:
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		word_length = 24;
+		break;
+
+	case SNDRV_PCM_FORMAT_U24_LE:
+	case SNDRV_PCM_FORMAT_S24_LE:
+		word_length = 24;
+		break;
+
+	case SNDRV_PCM_FORMAT_U32_LE:
+	case SNDRV_PCM_FORMAT_S32_LE:
+		word_length = 32;
+		break;
+
+	default:
+		printk(KERN_WARNING "davinci-mcasp: unsupported PCM format");
+		return -EINVAL;
+	}
+
 	ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt);
 	if (ret)
 		return ret;
@@ -1187,37 +1230,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_U8:
-	case SNDRV_PCM_FORMAT_S8:
-		word_length = 8;
-		break;
-
-	case SNDRV_PCM_FORMAT_U16_LE:
-	case SNDRV_PCM_FORMAT_S16_LE:
-		word_length = 16;
-		break;
-
-	case SNDRV_PCM_FORMAT_U24_3LE:
-	case SNDRV_PCM_FORMAT_S24_3LE:
-		word_length = 24;
-		break;
-
-	case SNDRV_PCM_FORMAT_U24_LE:
-	case SNDRV_PCM_FORMAT_S24_LE:
-		word_length = 24;
-		break;
-
-	case SNDRV_PCM_FORMAT_U32_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
-		word_length = 32;
-		break;
-
-	default:
-		printk(KERN_WARNING "davinci-mcasp: unsupported PCM format");
-		return -EINVAL;
-	}
-
 	davinci_config_channel_size(mcasp, word_length);
 
 	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE)
@@ -1404,12 +1416,13 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
-	 * limnit based on the seirializers * tdm_slots, we need to use that as
-	 * a constraint for the second stream.
-	 * Otherwise (first stream or less allowed channels) we use the
-	 * calculated constraint.
+	 * limit based on the seirializers * tdm_slots, and only one serializer
+	 * is in use we need to use that as a constraint for the second stream.
+	 * Otherwise (first stream or less allowed channels or more than one
+	 * serializer in use) we use the calculated constraint.
 	 */
-	if (mcasp->channels && mcasp->channels < max_channels)
+	if (mcasp->channels && mcasp->channels < max_channels &&
+	    ruledata->serializers == 1)
 		max_channels = mcasp->channels;
 	/*
 	 * But we can always allow channels upto the amount of
@@ -1470,6 +1483,7 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream,
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
 
 	mcasp->substreams[substream->stream] = NULL;
+	mcasp->active_serializers[substream->stream] = 0;
 
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
 		return;
-- 
cgit v1.2.3-59-g8ed1b


From 5e291a908869dd4fe70832a89678c6b0c1b8c481 Mon Sep 17 00:00:00 2001
From: Chuhong Yuan <hslester96@gmail.com>
Date: Thu, 25 Jul 2019 10:09:24 +0800
Subject: ALSA: echoaudio: Replace kmalloc + memcpy with kmemdup

Instead of using kmalloc + memcpy, use kmemdup
to simplify the code.

Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index b612a536a5a1..ca9125726be2 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2189,11 +2189,10 @@ static int snd_echo_resume(struct device *dev)
 	u32 pipe_alloc_mask;
 	int err;
 
-	commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL);
+	commpage = chip->comm_page;
+	commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL);
 	if (commpage_bak == NULL)
 		return -ENOMEM;
-	commpage = chip->comm_page;
-	memcpy(commpage_bak, commpage, sizeof(*commpage));
 
 	err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
 	if (err < 0) {
-- 
cgit v1.2.3-59-g8ed1b


From 5af9e972ac72b1c9f13cd35ccfb4687741c53f3e Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Thu, 25 Jul 2019 15:37:43 +0200
Subject: ASoC: SOF: Makefile - fix the top-level kernel module names (add snd-
 prefix)

Use the proper module name. The objs assignments are already there.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>0
Link: https://lore.kernel.org/r/20190725133743.22145-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/Makefile | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile
index 8f14c9d2950b..585fb6917489 100644
--- a/sound/soc/sof/Makefile
+++ b/sound/soc/sof/Makefile
@@ -11,8 +11,8 @@ obj-$(CONFIG_SND_SOC_SOF) += snd-sof.o
 obj-$(CONFIG_SND_SOC_SOF_NOCODEC) += snd-sof-nocodec.o
 
 
-obj-$(CONFIG_SND_SOC_SOF_ACPI) += sof-acpi-dev.o
-obj-$(CONFIG_SND_SOC_SOF_PCI) += sof-pci-dev.o
+obj-$(CONFIG_SND_SOC_SOF_ACPI) += snd-sof-acpi.o
+obj-$(CONFIG_SND_SOC_SOF_PCI) += snd-sof-pci.o
 
 obj-$(CONFIG_SND_SOC_SOF_INTEL_TOPLEVEL) += intel/
 obj-$(CONFIG_SND_SOC_SOF_XTENSA) += xtensa/
-- 
cgit v1.2.3-59-g8ed1b


From 01ec57a4371e573bfcfa898307af098a8c6f9dcf Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Thu, 25 Jul 2019 17:39:29 +0100
Subject: ASoC: wm_adsp: Allow bus error handler to be called directly

There is no need for end drivers to add helper functions to allow the
bus error handler to be called, simply update the prototype so it can be
called directly.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190725163931.24964-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs47l90.c | 9 +--------
 sound/soc/codecs/wm_adsp.c | 6 ++++--
 sound/soc/codecs/wm_adsp.h | 4 ++--
 3 files changed, 7 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c
index c4ecb0e6911a..67cac60a859d 100644
--- a/sound/soc/codecs/cs47l90.c
+++ b/sound/soc/codecs/cs47l90.c
@@ -2402,13 +2402,6 @@ static irqreturn_t cs47l90_adsp2_irq(int irq, void *data)
 	return IRQ_HANDLED;
 }
 
-static irqreturn_t cs47l90_dsp_bus_error(int irq, void *data)
-{
-	struct wm_adsp *dsp = (struct wm_adsp *)data;
-
-	return wm_adsp2_bus_error(dsp);
-}
-
 static int cs47l90_component_probe(struct snd_soc_component *component)
 {
 	struct cs47l90 *cs47l90 = snd_soc_component_get_drvdata(component);
@@ -2558,7 +2551,7 @@ static int cs47l90_probe(struct platform_device *pdev)
 
 		if (ret == 0) {
 			ret = madera_init_bus_error_irq(&cs47l90->core, i,
-							cs47l90_dsp_bus_error);
+							wm_adsp2_bus_error);
 			if (ret != 0)
 				wm_adsp2_remove(&cs47l90->core.adsp[i]);
 		}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f5fbadc5e7e2..ae28d9907c30 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -4242,8 +4242,9 @@ static void wm_adsp_fatal_error(struct wm_adsp *dsp)
 	}
 }
 
-irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp)
+irqreturn_t wm_adsp2_bus_error(int irq, void *data)
 {
+	struct wm_adsp *dsp = (struct wm_adsp *)data;
 	unsigned int val;
 	struct regmap *regmap = dsp->regmap;
 	int ret = 0;
@@ -4307,8 +4308,9 @@ error:
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_bus_error);
 
-irqreturn_t wm_halo_bus_error(struct wm_adsp *dsp)
+irqreturn_t wm_halo_bus_error(int irq, void *data)
 {
+	struct wm_adsp *dsp = (struct wm_adsp *)data;
 	struct regmap *regmap = dsp->regmap;
 	unsigned int fault[6];
 	struct reg_sequence clear[] = {
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index 3b03d1eb986f..aa634ef6c9f5 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -171,8 +171,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 int wm_adsp_early_event(struct snd_soc_dapm_widget *w,
 			struct snd_kcontrol *kcontrol, int event);
 
-irqreturn_t wm_adsp2_bus_error(struct wm_adsp *adsp);
-irqreturn_t wm_halo_bus_error(struct wm_adsp *dsp);
+irqreturn_t wm_adsp2_bus_error(int irq, void *data);
+irqreturn_t wm_halo_bus_error(int irq, void *data);
 irqreturn_t wm_halo_wdt_expire(int irq, void *data);
 
 int wm_adsp_event(struct snd_soc_dapm_widget *w,
-- 
cgit v1.2.3-59-g8ed1b


From 9cba2d6a144741ad99bfa379337bb984a8ed94bd Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Date: Thu, 25 Jul 2019 17:39:30 +0100
Subject: ASoC: cs47l15: Add codec driver for Cirrus Logic CS47L15

Adds the codec driver for the CS47L15 SmartCodec. This is a
multi-functional codec based on the Cirrus Logic Madera platform.

Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Jaswinder Jassal <jjassal@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190725163931.24964-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig   |    6 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/cs47l15.c | 1490 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/madera.c  |    4 +
 4 files changed, 1502 insertions(+)
 create mode 100644 sound/soc/codecs/cs47l15.c

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 9f89a5346299..dc23476d2eea 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -70,6 +70,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS43130 if I2C
 	select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CS4349 if I2C
+	select SND_SOC_CS47L15 if MFD_CS47L15
 	select SND_SOC_CS47L24 if MFD_CS47L24
 	select SND_SOC_CS47L35 if MFD_CS47L35
 	select SND_SOC_CS47L85 if MFD_CS47L85
@@ -581,6 +582,9 @@ config SND_SOC_CS4349
 	tristate "Cirrus Logic CS4349 CODEC"
 	depends on I2C
 
+config SND_SOC_CS47L15
+	tristate
+
 config SND_SOC_CS47L24
 	tristate
 
@@ -722,9 +726,11 @@ config SND_SOC_LOCHNAGAR_SC
 
 config SND_SOC_MADERA
 	tristate
+	default y if SND_SOC_CS47L15=y
 	default y if SND_SOC_CS47L35=y
 	default y if SND_SOC_CS47L85=y
 	default y if SND_SOC_CS47L90=y
+	default m if SND_SOC_CS47L15=m
 	default m if SND_SOC_CS47L35=m
 	default m if SND_SOC_CS47L85=m
 	default m if SND_SOC_CS47L90=m
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5b4bb8cf4325..174346b1418a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -64,6 +64,7 @@ snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
 snd-soc-cs43130-objs := cs43130.o
 snd-soc-cs4341-objs := cs4341.o
 snd-soc-cs4349-objs := cs4349.o
+snd-soc-cs47l15-objs := cs47l15.o
 snd-soc-cs47l24-objs := cs47l24.o
 snd-soc-cs47l35-objs := cs47l35.o
 snd-soc-cs47l85-objs := cs47l85.o
@@ -346,6 +347,7 @@ obj-$(CONFIG_SND_SOC_CS43130)   += snd-soc-cs43130.o
 obj-$(CONFIG_SND_SOC_CS4341)	+= snd-soc-cs4341.o
 obj-$(CONFIG_SND_SOC_CS4349)	+= snd-soc-cs4349.o
 obj-$(CONFIG_SND_SOC_CS47L24)	+= snd-soc-cs47l24.o
+obj-$(CONFIG_SND_SOC_CS47L15)	+= snd-soc-cs47l15.o
 obj-$(CONFIG_SND_SOC_CS47L35)	+= snd-soc-cs47l35.o
 obj-$(CONFIG_SND_SOC_CS47L85)	+= snd-soc-cs47l85.o
 obj-$(CONFIG_SND_SOC_CS47L90)	+= snd-soc-cs47l90.o
diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c
new file mode 100644
index 000000000000..ece1276f38eb
--- /dev/null
+++ b/sound/soc/codecs/cs47l15.c
@@ -0,0 +1,1490 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// ALSA SoC Audio driver for CS47L15 codec
+//
+// Copyright (C) 2016-2019 Cirrus Logic, Inc. and
+//                         Cirrus Logic International Semiconductor Ltd.
+//
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include <linux/irqchip/irq-madera.h>
+#include <linux/mfd/madera/core.h>
+#include <linux/mfd/madera/registers.h>
+
+#include "madera.h"
+#include "wm_adsp.h"
+
+#define CS47L15_NUM_ADSP 1
+#define CS47L15_MONO_OUTPUTS 1
+
+/* Mid-mode registers */
+#define CS47L15_ADC_INT_BIAS_MASK	0x3800
+#define CS47L15_ADC_INT_BIAS_SHIFT	11
+#define CS47L15_PGA_BIAS_SEL_MASK	0x03
+#define CS47L15_PGA_BIAS_SEL_SHIFT	0
+
+#define DRV_NAME "cs47l15-codec"
+
+struct cs47l15 {
+	struct madera_priv core;
+	struct madera_fll fll[2];
+
+	bool in1_lp_mode;
+};
+
+static const struct wm_adsp_region cs47l15_dsp1_regions[] = {
+	{ .type = WMFW_ADSP2_PM, .base = 0x080000 },
+	{ .type = WMFW_ADSP2_ZM, .base = 0x0e0000 },
+	{ .type = WMFW_ADSP2_XM, .base = 0x0a0000 },
+	{ .type = WMFW_ADSP2_YM, .base = 0x0c0000 },
+};
+
+static const char * const cs47l15_outdemux_texts[] = {
+	"HPOUT",
+	"EPOUT",
+};
+
+static SOC_ENUM_SINGLE_DECL(cs47l15_outdemux_enum, SND_SOC_NOPM, 0,
+			    cs47l15_outdemux_texts);
+
+static const struct snd_kcontrol_new cs47l15_outdemux =
+	SOC_DAPM_ENUM_EXT("HPOUT1 Demux", cs47l15_outdemux_enum,
+			  madera_out1_demux_get, madera_out1_demux_put);
+
+static int cs47l15_adsp_power_ev(struct snd_soc_dapm_widget *w,
+				 struct snd_kcontrol *kcontrol,
+				 int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component);
+	struct madera_priv *priv = &cs47l15->core;
+	struct madera *madera = priv->madera;
+	unsigned int freq;
+	int ret;
+
+	ret = regmap_read(madera->regmap, MADERA_DSP_CLOCK_2, &freq);
+	if (ret != 0) {
+		dev_err(madera->dev,
+			"Failed to read MADERA_DSP_CLOCK_2: %d\n", ret);
+		return ret;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		ret = madera_set_adsp_clk(&cs47l15->core, w->shift, freq);
+		if (ret)
+			return ret;
+		break;
+	default:
+		break;
+	}
+
+	return wm_adsp_early_event(w, kcontrol, event);
+}
+
+#define CS47L15_NG_SRC(name, base) \
+	SOC_SINGLE(name " NG HPOUT1L Switch",  base,  0, 1, 0), \
+	SOC_SINGLE(name " NG HPOUT1R Switch",  base,  1, 1, 0), \
+	SOC_SINGLE(name " NG SPKOUTL Switch",  base,  6, 1, 0), \
+	SOC_SINGLE(name " NG SPKDAT1L Switch", base,  8, 1, 0), \
+	SOC_SINGLE(name " NG SPKDAT1R Switch", base,  9, 1, 0)
+
+static int cs47l15_in1_adc_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+		snd_soc_kcontrol_component(kcontrol);
+	struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component);
+
+	ucontrol->value.integer.value[0] = !!cs47l15->in1_lp_mode;
+
+	return 0;
+}
+
+static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+		snd_soc_kcontrol_component(kcontrol);
+	struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component);
+
+	switch (ucontrol->value.integer.value[0]) {
+	case 0:
+		/* Set IN1 to normal mode */
+		snd_soc_component_update_bits(component, MADERA_DMIC1L_CONTROL,
+					      MADERA_IN1_OSR_MASK,
+					      5 << MADERA_IN1_OSR_SHIFT);
+		snd_soc_component_update_bits(component, CS47L15_ADC_INT_BIAS,
+					      CS47L15_ADC_INT_BIAS_MASK,
+					      4 << CS47L15_ADC_INT_BIAS_SHIFT);
+		snd_soc_component_update_bits(component, CS47L15_PGA_BIAS_SEL,
+					      CS47L15_PGA_BIAS_SEL_MASK, 0);
+		cs47l15->in1_lp_mode = false;
+		break;
+	default:
+		/* Set IN1 to LP mode */
+		snd_soc_component_update_bits(component, MADERA_DMIC1L_CONTROL,
+					      MADERA_IN1_OSR_MASK,
+					      4 << MADERA_IN1_OSR_SHIFT);
+		snd_soc_component_update_bits(component, CS47L15_ADC_INT_BIAS,
+					      CS47L15_ADC_INT_BIAS_MASK,
+					      1 << CS47L15_ADC_INT_BIAS_SHIFT);
+		snd_soc_component_update_bits(component, CS47L15_PGA_BIAS_SEL,
+					      CS47L15_PGA_BIAS_SEL_MASK,
+					      3 << CS47L15_PGA_BIAS_SEL_SHIFT);
+		cs47l15->in1_lp_mode = true;
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new cs47l15_snd_controls[] = {
+SOC_ENUM("IN1 OSR", madera_in_dmic_osr[0]),
+SOC_ENUM("IN2 OSR", madera_in_dmic_osr[1]),
+
+SOC_SINGLE_RANGE_TLV("IN1L Volume", MADERA_IN1L_CONTROL,
+		     MADERA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN1R Volume", MADERA_IN1R_CONTROL,
+		     MADERA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv),
+
+SOC_ENUM("IN HPF Cutoff Frequency", madera_in_hpf_cut_enum),
+
+SOC_SINGLE("IN1L HPF Switch", MADERA_IN1L_CONTROL, MADERA_IN1L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN1R HPF Switch", MADERA_IN1R_CONTROL, MADERA_IN1R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2L HPF Switch", MADERA_IN2L_CONTROL, MADERA_IN2L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2R HPF Switch", MADERA_IN2R_CONTROL, MADERA_IN2R_HPF_SHIFT, 1, 0),
+
+SOC_SINGLE_TLV("IN1L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1L,
+	       MADERA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN1R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1R,
+	       MADERA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN2L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2L,
+	       MADERA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN2R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2R,
+	       MADERA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+
+SOC_ENUM("Input Ramp Up", madera_in_vi_ramp),
+SOC_ENUM("Input Ramp Down", madera_in_vd_ramp),
+
+MADERA_MIXER_CONTROLS("EQ1", MADERA_EQ1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("EQ2", MADERA_EQ2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("EQ3", MADERA_EQ3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("EQ4", MADERA_EQ4MIX_INPUT_1_SOURCE),
+
+MADERA_EQ_CONTROL("EQ1 Coefficients", MADERA_EQ1_2),
+SOC_SINGLE_TLV("EQ1 B1 Volume", MADERA_EQ1_1, MADERA_EQ1_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", MADERA_EQ1_1, MADERA_EQ1_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", MADERA_EQ1_1, MADERA_EQ1_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", MADERA_EQ1_2, MADERA_EQ1_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", MADERA_EQ1_2, MADERA_EQ1_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+MADERA_EQ_CONTROL("EQ2 Coefficients", MADERA_EQ2_2),
+SOC_SINGLE_TLV("EQ2 B1 Volume", MADERA_EQ2_1, MADERA_EQ2_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", MADERA_EQ2_1, MADERA_EQ2_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", MADERA_EQ2_1, MADERA_EQ2_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", MADERA_EQ2_2, MADERA_EQ2_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", MADERA_EQ2_2, MADERA_EQ2_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+MADERA_EQ_CONTROL("EQ3 Coefficients", MADERA_EQ3_2),
+SOC_SINGLE_TLV("EQ3 B1 Volume", MADERA_EQ3_1, MADERA_EQ3_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", MADERA_EQ3_1, MADERA_EQ3_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", MADERA_EQ3_1, MADERA_EQ3_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", MADERA_EQ3_2, MADERA_EQ3_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", MADERA_EQ3_2, MADERA_EQ3_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+MADERA_EQ_CONTROL("EQ4 Coefficients", MADERA_EQ4_2),
+SOC_SINGLE_TLV("EQ4 B1 Volume", MADERA_EQ4_1, MADERA_EQ4_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", MADERA_EQ4_1, MADERA_EQ4_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", MADERA_EQ4_1, MADERA_EQ4_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", MADERA_EQ4_2, MADERA_EQ4_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", MADERA_EQ4_2, MADERA_EQ4_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+MADERA_MIXER_CONTROLS("DRC1L", MADERA_DRC1LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DRC1R", MADERA_DRC1RMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DRC2L", MADERA_DRC2LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DRC2R", MADERA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", MADERA_DRC1_CTRL1, 5,
+		   MADERA_DRC1R_ENA | MADERA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", MADERA_DRC2_CTRL1, 5,
+		   MADERA_DRC2R_ENA | MADERA_DRC2L_ENA),
+
+MADERA_MIXER_CONTROLS("LHPF1", MADERA_HPLP1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("LHPF2", MADERA_HPLP2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("LHPF3", MADERA_HPLP3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("LHPF4", MADERA_HPLP4MIX_INPUT_1_SOURCE),
+
+MADERA_LHPF_CONTROL("LHPF1 Coefficients", MADERA_HPLPF1_2),
+MADERA_LHPF_CONTROL("LHPF2 Coefficients", MADERA_HPLPF2_2),
+MADERA_LHPF_CONTROL("LHPF3 Coefficients", MADERA_HPLPF3_2),
+MADERA_LHPF_CONTROL("LHPF4 Coefficients", MADERA_HPLPF4_2),
+
+SOC_ENUM("LHPF1 Mode", madera_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", madera_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", madera_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", madera_lhpf4_mode),
+
+MADERA_RATE_ENUM("ISRC1 FSL", madera_isrc_fsl[0]),
+MADERA_RATE_ENUM("ISRC2 FSL", madera_isrc_fsl[1]),
+MADERA_RATE_ENUM("ISRC1 FSH", madera_isrc_fsh[0]),
+MADERA_RATE_ENUM("ISRC2 FSH", madera_isrc_fsh[1]),
+
+WM_ADSP2_PRELOAD_SWITCH("DSP1", 1),
+
+MADERA_MIXER_CONTROLS("DSP1L", MADERA_DSP1LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DSP1R", MADERA_DSP1RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", MADERA_COMFORT_NOISE_GENERATOR,
+	       MADERA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, madera_noise_tlv),
+
+MADERA_MIXER_CONTROLS("HPOUT1L", MADERA_OUT1LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("HPOUT1R", MADERA_OUT1RMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SPKOUTL", MADERA_OUT4LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SPKDAT1L", MADERA_OUT5LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SPKDAT1R", MADERA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 SC Protect Switch", MADERA_HP1_SHORT_CIRCUIT_CTRL,
+	   MADERA_HP1_SC_ENA_SHIFT, 1, 0),
+
+SOC_SINGLE("SPKDAT1 High Performance Switch", MADERA_OUTPUT_PATH_CONFIG_5L,
+	   MADERA_OUT5_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_1L,
+	     MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("Speaker Digital Switch", MADERA_DAC_DIGITAL_VOLUME_4L,
+	   MADERA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_5L,
+	     MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_1L,
+		 MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("Speaker Digital Volume", MADERA_DAC_DIGITAL_VOLUME_4L,
+	       MADERA_OUT4L_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_5L,
+		 MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, madera_digital_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", MADERA_PDM_SPK1_CTRL_1, MADERA_SPK1L_MUTE_SHIFT,
+	   MADERA_SPK1R_MUTE_SHIFT, 1, 1),
+
+SOC_ENUM("Output Ramp Up", madera_out_vi_ramp),
+SOC_ENUM("Output Ramp Down", madera_out_vd_ramp),
+
+SOC_SINGLE("Noise Gate Switch", MADERA_NOISE_GATE_CONTROL,
+	   MADERA_NGATE_ENA_SHIFT, 1, 0),
+SOC_SINGLE_TLV("Noise Gate Threshold Volume", MADERA_NOISE_GATE_CONTROL,
+	       MADERA_NGATE_THR_SHIFT, 7, 1, madera_ng_tlv),
+SOC_ENUM("Noise Gate Hold", madera_ng_hold),
+
+SOC_SINGLE_BOOL_EXT("IN1 LP Mode Switch", 0,
+		    cs47l15_in1_adc_get, cs47l15_in1_adc_put),
+
+CS47L15_NG_SRC("HPOUT1L", MADERA_NOISE_GATE_SELECT_1L),
+CS47L15_NG_SRC("HPOUT1R", MADERA_NOISE_GATE_SELECT_1R),
+CS47L15_NG_SRC("SPKOUTL", MADERA_NOISE_GATE_SELECT_4L),
+CS47L15_NG_SRC("SPKDAT1L", MADERA_NOISE_GATE_SELECT_5L),
+CS47L15_NG_SRC("SPKDAT1R", MADERA_NOISE_GATE_SELECT_5R),
+
+MADERA_MIXER_CONTROLS("AIF1TX1", MADERA_AIF1TX1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX2", MADERA_AIF1TX2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX3", MADERA_AIF1TX3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX4", MADERA_AIF1TX4MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX5", MADERA_AIF1TX5MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX6", MADERA_AIF1TX6MIX_INPUT_1_SOURCE),
+
+MADERA_MIXER_CONTROLS("AIF2TX1", MADERA_AIF2TX1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX2", MADERA_AIF2TX2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX3", MADERA_AIF2TX3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX4", MADERA_AIF2TX4MIX_INPUT_1_SOURCE),
+
+MADERA_MIXER_CONTROLS("AIF3TX1", MADERA_AIF3TX1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF3TX2", MADERA_AIF3TX2MIX_INPUT_1_SOURCE),
+
+MADERA_GAINMUX_CONTROLS("SPDIF1TX1", MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE),
+MADERA_GAINMUX_CONTROLS("SPDIF1TX2", MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+};
+
+MADERA_MIXER_ENUMS(EQ1, MADERA_EQ1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(EQ2, MADERA_EQ2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(EQ3, MADERA_EQ3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(EQ4, MADERA_EQ4MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(DRC1L, MADERA_DRC1LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DRC1R, MADERA_DRC1RMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DRC2L, MADERA_DRC2LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DRC2R, MADERA_DRC2RMIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(LHPF1, MADERA_HPLP1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(LHPF2, MADERA_HPLP2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(LHPF3, MADERA_HPLP3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(LHPF4, MADERA_HPLP4MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(DSP1L, MADERA_DSP1LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DSP1R, MADERA_DSP1RMIX_INPUT_1_SOURCE);
+MADERA_DSP_AUX_ENUMS(DSP1, MADERA_DSP1AUX1MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(PWM1, MADERA_PWM1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(PWM2, MADERA_PWM2MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(OUT1L, MADERA_OUT1LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(OUT1R, MADERA_OUT1RMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SPKOUTL, MADERA_OUT4LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SPKDAT1L, MADERA_OUT5LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SPKDAT1R, MADERA_OUT5RMIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(AIF1TX1, MADERA_AIF1TX1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX2, MADERA_AIF1TX2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX3, MADERA_AIF1TX3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX4, MADERA_AIF1TX4MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX5, MADERA_AIF1TX5MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX6, MADERA_AIF1TX6MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(AIF2TX1, MADERA_AIF2TX1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX2, MADERA_AIF2TX2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX3, MADERA_AIF2TX3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX4, MADERA_AIF2TX4MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(AIF3TX1, MADERA_AIF3TX1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF3TX2, MADERA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(SPD1TX1, MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(SPD1TX2, MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC1INT1, MADERA_ISRC1INT1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1INT2, MADERA_ISRC1INT2MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1INT3, MADERA_ISRC1INT3MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1INT4, MADERA_ISRC1INT4MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC1DEC1, MADERA_ISRC1DEC1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1DEC2, MADERA_ISRC1DEC2MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1DEC3, MADERA_ISRC1DEC3MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1DEC4, MADERA_ISRC1DEC4MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC2INT1, MADERA_ISRC2INT1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2INT2, MADERA_ISRC2INT2MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2INT3, MADERA_ISRC2INT3MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2INT4, MADERA_ISRC2INT4MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC2DEC1, MADERA_ISRC2DEC1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2DEC2, MADERA_ISRC2DEC2MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2DEC3, MADERA_ISRC2DEC3MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2DEC4, MADERA_ISRC2DEC4MIX_INPUT_1_SOURCE);
+
+static const char * const cs47l15_aec_loopback_texts[] = {
+	"HPOUT1L", "HPOUT1R", "SPKOUTL", "SPKDAT1L", "SPKDAT1R",
+};
+
+static const unsigned int cs47l15_aec_loopback_values[] = {
+	0, 1, 6, 8, 9,
+};
+
+static const struct soc_enum cs47l15_aec1_loopback =
+	SOC_VALUE_ENUM_SINGLE(MADERA_DAC_AEC_CONTROL_1,
+			      MADERA_AEC1_LOOPBACK_SRC_SHIFT, 0xf,
+			      ARRAY_SIZE(cs47l15_aec_loopback_texts),
+			      cs47l15_aec_loopback_texts,
+			      cs47l15_aec_loopback_values);
+
+static const struct soc_enum cs47l15_aec2_loopback =
+	SOC_VALUE_ENUM_SINGLE(MADERA_DAC_AEC_CONTROL_2,
+			      MADERA_AEC2_LOOPBACK_SRC_SHIFT, 0xf,
+			      ARRAY_SIZE(cs47l15_aec_loopback_texts),
+			      cs47l15_aec_loopback_texts,
+			      cs47l15_aec_loopback_values);
+
+static const struct snd_kcontrol_new cs47l15_aec_loopback_mux[] = {
+	SOC_DAPM_ENUM("AEC1 Loopback", cs47l15_aec1_loopback),
+	SOC_DAPM_ENUM("AEC2 Loopback", cs47l15_aec2_loopback),
+};
+
+static const struct snd_soc_dapm_widget cs47l15_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT,
+		    0, madera_sysclk_ev,
+		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK,
+		    MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1,
+		    MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD1", 20, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", MADERA_MIC_BIAS_CTRL_1,
+		    MADERA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1A", MADERA_MIC_BIAS_CTRL_5,
+		    MADERA_MICB1A_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS1B", MADERA_MIC_BIAS_CTRL_5,
+		    MADERA_MICB1B_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS1C", MADERA_MIC_BIAS_CTRL_5,
+		    MADERA_MICB1C_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("FXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_FX, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("ISRC1CLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_ISRC1, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("ISRC2CLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_ISRC2, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("OUTCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_OUT, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("SPDCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_SPD, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_DSP1, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF1TXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_AIF1, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF2TXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_AIF2, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF3TXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_AIF3, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("PWMCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_PWM, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1ALN"),
+SND_SOC_DAPM_INPUT("IN1ALP"),
+SND_SOC_DAPM_INPUT("IN1BLN"),
+SND_SOC_DAPM_INPUT("IN1BLP"),
+SND_SOC_DAPM_INPUT("IN1ARN"),
+SND_SOC_DAPM_INPUT("IN1ARP"),
+SND_SOC_DAPM_INPUT("IN1BRN"),
+SND_SOC_DAPM_INPUT("IN1BRP"),
+SND_SOC_DAPM_INPUT("IN2N"),
+SND_SOC_DAPM_INPUT("IN2P"),
+SND_SOC_DAPM_INPUT("SPKRXDAT"),
+
+SND_SOC_DAPM_MUX("IN1L Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[0]),
+SND_SOC_DAPM_MUX("IN1R Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[1]),
+
+SND_SOC_DAPM_MUX("IN1L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]),
+SND_SOC_DAPM_MUX("IN1R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]),
+
+SND_SOC_DAPM_MUX("IN2L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]),
+SND_SOC_DAPM_MUX("IN2R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]),
+
+SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
+SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
+
+SND_SOC_DAPM_OUTPUT("DSP Trigger Out"),
+
+SND_SOC_DAPM_DEMUX("HPOUT1 Demux", SND_SOC_NOPM, 0, 0, &cs47l15_outdemux),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM1_ENA_SHIFT,
+		 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM2_ENA_SHIFT,
+		 0, NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX4_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+		     MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+		     MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
+		   MADERA_OUT1L_ENA_SHIFT, 0, NULL, 0, madera_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
+		   MADERA_OUT1R_ENA_SHIFT, 0, NULL, 0, madera_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4L", SND_SOC_NOPM,
+		   MADERA_OUT4L_ENA_SHIFT, 0, NULL, 0, madera_spk_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", MADERA_OUTPUT_ENABLES_1,
+		   MADERA_OUT5L_ENA_SHIFT, 0, NULL, 0, madera_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", MADERA_OUTPUT_ENABLES_1,
+		   MADERA_OUT5R_ENA_SHIFT, 0, NULL, 0, madera_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_PGA("SPD1TX1", MADERA_SPD1_TX_CONTROL,
+		 MADERA_SPD1_VAL1_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SPD1TX2", MADERA_SPD1_TX_CONTROL,
+		 MADERA_SPD1_VAL2_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("SPD1", MADERA_SPD1_TX_CONTROL,
+		     MADERA_SPD1_ENA_SHIFT, 0, NULL, 0),
+
+/*
+ * mux_in widgets : arranged in the order of sources
+ * specified in MADERA_MIXER_INPUT_ROUTES
+ */
+
+SND_SOC_DAPM_PGA("Noise Generator", MADERA_COMFORT_NOISE_GENERATOR,
+		 MADERA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", MADERA_TONE_GENERATOR_1,
+		 MADERA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", MADERA_TONE_GENERATOR_1,
+		 MADERA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_SIGGEN("HAPTICS"),
+
+SND_SOC_DAPM_MUX("AEC1 Loopback", MADERA_DAC_AEC_CONTROL_1,
+		 MADERA_AEC1_LOOPBACK_ENA_SHIFT, 0,
+		 &cs47l15_aec_loopback_mux[0]),
+SND_SOC_DAPM_MUX("AEC2 Loopback", MADERA_DAC_AEC_CONTROL_2,
+		 MADERA_AEC2_LOOPBACK_ENA_SHIFT, 0,
+		 &cs47l15_aec_loopback_mux[1]),
+
+SND_SOC_DAPM_PGA_E("IN1L", MADERA_INPUT_ENABLES, MADERA_IN1L_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R", MADERA_INPUT_ENABLES, MADERA_IN1R_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L", MADERA_INPUT_ENABLES, MADERA_IN2L_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R", MADERA_INPUT_ENABLES, MADERA_IN2R_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX4_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+		    MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+		    MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA("EQ1", MADERA_EQ1_1, MADERA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", MADERA_EQ2_1, MADERA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", MADERA_EQ3_1, MADERA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", MADERA_EQ4_1, MADERA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", MADERA_DRC1_CTRL1, MADERA_DRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", MADERA_DRC1_CTRL1, MADERA_DRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", MADERA_DRC2_CTRL1, MADERA_DRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", MADERA_DRC2_CTRL1, MADERA_DRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", MADERA_HPLPF1_1, MADERA_LHPF1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", MADERA_HPLPF2_1, MADERA_LHPF2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", MADERA_HPLPF3_1, MADERA_LHPF3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", MADERA_HPLPF4_1, MADERA_LHPF4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1DEC1", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC2", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC3", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC4", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_DEC4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1INT1", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT2", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT3", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT4", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_INT4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2DEC1", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC2", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC3", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_DEC3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC4", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_DEC4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2INT1", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT2", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT3", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_INT3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT4", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_INT4_ENA_SHIFT, 0, NULL, 0),
+
+WM_ADSP2("DSP1", 0, cs47l15_adsp_power_ev),
+
+/* end of ordered widget list */
+
+MADERA_MIXER_WIDGETS(EQ1, "EQ1"),
+MADERA_MIXER_WIDGETS(EQ2, "EQ2"),
+MADERA_MIXER_WIDGETS(EQ3, "EQ3"),
+MADERA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+MADERA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+MADERA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+MADERA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+MADERA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+SND_SOC_DAPM_SWITCH("DRC1 Activity Output", SND_SOC_NOPM, 0, 0,
+		    &madera_drc_activity_output_mux[0]),
+SND_SOC_DAPM_SWITCH("DRC2 Activity Output", SND_SOC_NOPM, 0, 0,
+		    &madera_drc_activity_output_mux[1]),
+
+MADERA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+MADERA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+MADERA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+MADERA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+MADERA_MIXER_WIDGETS(PWM1, "PWM1"),
+MADERA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+MADERA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+MADERA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+MADERA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+MADERA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+MADERA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+
+MADERA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+MADERA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+MADERA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+MADERA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+MADERA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+MADERA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+
+MADERA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+MADERA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+MADERA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"),
+MADERA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"),
+
+MADERA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+MADERA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+MADERA_MUX_WIDGETS(SPD1TX1, "SPDIF1TX1"),
+MADERA_MUX_WIDGETS(SPD1TX2, "SPDIF1TX2"),
+
+MADERA_DSP_WIDGETS(DSP1, "DSP1"),
+
+SND_SOC_DAPM_SWITCH("DSP1 Trigger Output", SND_SOC_NOPM, 0, 0,
+		    &madera_dsp_trigger_output_mux[0]),
+
+MADERA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"),
+MADERA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"),
+MADERA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"),
+MADERA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"),
+
+MADERA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"),
+MADERA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"),
+MADERA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"),
+MADERA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"),
+
+MADERA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"),
+MADERA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
+MADERA_MUX_WIDGETS(ISRC2DEC3, "ISRC2DEC3"),
+MADERA_MUX_WIDGETS(ISRC2DEC4, "ISRC2DEC4"),
+
+MADERA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
+MADERA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
+MADERA_MUX_WIDGETS(ISRC2INT3, "ISRC2INT3"),
+MADERA_MUX_WIDGETS(ISRC2INT4, "ISRC2INT4"),
+
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+SND_SOC_DAPM_OUTPUT("SPDIF1"),
+
+SND_SOC_DAPM_OUTPUT("MICSUPP"),
+};
+
+#define MADERA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Noise Generator", "Noise Generator" }, \
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "Haptics", "HAPTICS" }, \
+	{ name, "AEC1", "AEC1 Loopback" }, \
+	{ name, "AEC2", "AEC2 Loopback" }, \
+	{ name, "IN1L", "IN1L" }, \
+	{ name, "IN1R", "IN1R" }, \
+	{ name, "IN2L", "IN2L" }, \
+	{ name, "IN2R", "IN2R" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF2RX3", "AIF2RX3" }, \
+	{ name, "AIF2RX4", "AIF2RX4" }, \
+	{ name, "AIF3RX1", "AIF3RX1" }, \
+	{ name, "AIF3RX2", "AIF3RX2" }, \
+	{ name, "EQ1", "EQ1" }, \
+	{ name, "EQ2", "EQ2" }, \
+	{ name, "EQ3", "EQ3" }, \
+	{ name, "EQ4", "EQ4" }, \
+	{ name, "DRC1L", "DRC1L" }, \
+	{ name, "DRC1R", "DRC1R" }, \
+	{ name, "DRC2L", "DRC2L" }, \
+	{ name, "DRC2R", "DRC2R" }, \
+	{ name, "LHPF1", "LHPF1" }, \
+	{ name, "LHPF2", "LHPF2" }, \
+	{ name, "LHPF3", "LHPF3" }, \
+	{ name, "LHPF4", "LHPF4" }, \
+	{ name, "ISRC1DEC1", "ISRC1DEC1" }, \
+	{ name, "ISRC1DEC2", "ISRC1DEC2" }, \
+	{ name, "ISRC1DEC3", "ISRC1DEC3" }, \
+	{ name, "ISRC1DEC4", "ISRC1DEC4" }, \
+	{ name, "ISRC1INT1", "ISRC1INT1" }, \
+	{ name, "ISRC1INT2", "ISRC1INT2" }, \
+	{ name, "ISRC1INT3", "ISRC1INT3" }, \
+	{ name, "ISRC1INT4", "ISRC1INT4" }, \
+	{ name, "ISRC2DEC1", "ISRC2DEC1" }, \
+	{ name, "ISRC2DEC2", "ISRC2DEC2" }, \
+	{ name, "ISRC2DEC3", "ISRC2DEC3" }, \
+	{ name, "ISRC2DEC4", "ISRC2DEC4" }, \
+	{ name, "ISRC2INT1", "ISRC2INT1" }, \
+	{ name, "ISRC2INT2", "ISRC2INT2" }, \
+	{ name, "ISRC2INT3", "ISRC2INT3" }, \
+	{ name, "ISRC2INT4", "ISRC2INT4" }, \
+	{ name, "DSP1.1", "DSP1" }, \
+	{ name, "DSP1.2", "DSP1" }, \
+	{ name, "DSP1.3", "DSP1" }, \
+	{ name, "DSP1.4", "DSP1" }, \
+	{ name, "DSP1.5", "DSP1" }, \
+	{ name, "DSP1.6", "DSP1" }
+
+static const struct snd_soc_dapm_route cs47l15_dapm_routes[] = {
+	/* Internal clock domains */
+	{ "EQ1", NULL, "FXCLK" },
+	{ "EQ2", NULL, "FXCLK" },
+	{ "EQ3", NULL, "FXCLK" },
+	{ "EQ4", NULL, "FXCLK" },
+	{ "DRC1L", NULL, "FXCLK" },
+	{ "DRC1R", NULL, "FXCLK" },
+	{ "DRC2L", NULL, "FXCLK" },
+	{ "DRC2R", NULL, "FXCLK" },
+	{ "LHPF1", NULL, "FXCLK" },
+	{ "LHPF2", NULL, "FXCLK" },
+	{ "LHPF3", NULL, "FXCLK" },
+	{ "LHPF4", NULL, "FXCLK" },
+	{ "PWM1 Mixer", NULL, "PWMCLK" },
+	{ "PWM2 Mixer", NULL, "PWMCLK" },
+	{ "OUT1L", NULL, "OUTCLK" },
+	{ "OUT1R", NULL, "OUTCLK" },
+	{ "OUT4L", NULL, "OUTCLK" },
+	{ "OUT5L", NULL, "OUTCLK" },
+	{ "OUT5R", NULL, "OUTCLK" },
+	{ "AIF1TX1", NULL, "AIF1TXCLK" },
+	{ "AIF1TX2", NULL, "AIF1TXCLK" },
+	{ "AIF1TX3", NULL, "AIF1TXCLK" },
+	{ "AIF1TX4", NULL, "AIF1TXCLK" },
+	{ "AIF1TX5", NULL, "AIF1TXCLK" },
+	{ "AIF1TX6", NULL, "AIF1TXCLK" },
+	{ "AIF2TX1", NULL, "AIF2TXCLK" },
+	{ "AIF2TX2", NULL, "AIF2TXCLK" },
+	{ "AIF2TX3", NULL, "AIF2TXCLK" },
+	{ "AIF2TX4", NULL, "AIF2TXCLK" },
+	{ "AIF3TX1", NULL, "AIF3TXCLK" },
+	{ "AIF3TX2", NULL, "AIF3TXCLK" },
+	{ "SPD1TX1", NULL, "SPDCLK" },
+	{ "SPD1TX2", NULL, "SPDCLK" },
+	{ "DSP1", NULL, "DSP1CLK" },
+	{ "ISRC1DEC1", NULL, "ISRC1CLK" },
+	{ "ISRC1DEC2", NULL, "ISRC1CLK" },
+	{ "ISRC1DEC3", NULL, "ISRC1CLK" },
+	{ "ISRC1DEC4", NULL, "ISRC1CLK" },
+	{ "ISRC1INT1", NULL, "ISRC1CLK" },
+	{ "ISRC1INT2", NULL, "ISRC1CLK" },
+	{ "ISRC1INT3", NULL, "ISRC1CLK" },
+	{ "ISRC1INT4", NULL, "ISRC1CLK" },
+	{ "ISRC2DEC1", NULL, "ISRC2CLK" },
+	{ "ISRC2DEC2", NULL, "ISRC2CLK" },
+	{ "ISRC2DEC3", NULL, "ISRC2CLK" },
+	{ "ISRC2DEC4", NULL, "ISRC2CLK" },
+	{ "ISRC2INT1", NULL, "ISRC2CLK" },
+	{ "ISRC2INT2", NULL, "ISRC2CLK" },
+	{ "ISRC2INT3", NULL, "ISRC2CLK" },
+	{ "ISRC2INT4", NULL, "ISRC2CLK" },
+
+	{ "OUT1L", NULL, "CPVDD1" },
+	{ "OUT1R", NULL, "CPVDD1" },
+	{ "OUT4L", NULL, "SPKVDD" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT4L", NULL, "SYSCLK" },
+	{ "OUT5L", NULL, "SYSCLK" },
+	{ "OUT5R", NULL, "SYSCLK" },
+
+	{ "SPD1", NULL, "SYSCLK" },
+	{ "SPD1", NULL, "SPD1TX1" },
+	{ "SPD1", NULL, "SPD1TX2" },
+
+	{ "IN1L", NULL, "SYSCLK" },
+	{ "IN1R", NULL, "SYSCLK" },
+	{ "IN2L", NULL, "SYSCLK" },
+	{ "IN2R", NULL, "SYSCLK" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+
+	{ "MICBIAS1A", NULL, "MICBIAS1" },
+	{ "MICBIAS1B", NULL, "MICBIAS1" },
+	{ "MICBIAS1C", NULL, "MICBIAS1" },
+
+	{ "Noise Generator", NULL, "SYSCLK" },
+	{ "Tone Generator 1", NULL, "SYSCLK" },
+	{ "Tone Generator 2", NULL, "SYSCLK" },
+
+	{ "Noise Generator", NULL, "NOISE" },
+	{ "Tone Generator 1", NULL, "TONE" },
+	{ "Tone Generator 2", NULL, "TONE" },
+
+	{ "AIF1 Capture", NULL, "AIF1TX1" },
+	{ "AIF1 Capture", NULL, "AIF1TX2" },
+	{ "AIF1 Capture", NULL, "AIF1TX3" },
+	{ "AIF1 Capture", NULL, "AIF1TX4" },
+	{ "AIF1 Capture", NULL, "AIF1TX5" },
+	{ "AIF1 Capture", NULL, "AIF1TX6" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+	{ "AIF2 Capture", NULL, "AIF2TX3" },
+	{ "AIF2 Capture", NULL, "AIF2TX4" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+	{ "AIF2RX3", NULL, "AIF2 Playback" },
+	{ "AIF2RX4", NULL, "AIF2 Playback" },
+
+	{ "AIF3 Capture", NULL, "AIF3TX1" },
+	{ "AIF3 Capture", NULL, "AIF3TX2" },
+
+	{ "AIF3RX1", NULL, "AIF3 Playback" },
+	{ "AIF3RX2", NULL, "AIF3 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "AIF3 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "AIF3 Capture", NULL, "SYSCLK" },
+
+	{ "Audio Trace DSP", NULL, "DSP1" },
+
+	{ "IN1L Analog Mux", "A", "IN1ALN" },
+	{ "IN1L Analog Mux", "A", "IN1ALP" },
+	{ "IN1L Analog Mux", "B", "IN1BLN" },
+	{ "IN1L Analog Mux", "B", "IN1BLP" },
+	{ "IN1R Analog Mux", "A", "IN1ARN" },
+	{ "IN1R Analog Mux", "A", "IN1ARP" },
+	{ "IN1R Analog Mux", "B", "IN1BRN" },
+	{ "IN1R Analog Mux", "B", "IN1BRP" },
+
+	{ "IN1L Mode", "Analog", "IN1L Analog Mux" },
+	{ "IN1R Mode", "Analog", "IN1R Analog Mux" },
+
+	{ "IN1L Mode", "Digital", "IN1ALN" },
+	{ "IN1L Mode", "Digital", "IN1ALP" },
+	{ "IN1R Mode", "Digital", "IN1ALN" },
+	{ "IN1R Mode", "Digital", "IN1ALP" },
+
+	{ "IN1L", NULL, "IN1L Mode" },
+	{ "IN1R", NULL, "IN1R Mode" },
+
+	{ "IN2L Mode", "Analog", "IN2N" },
+	{ "IN2L Mode", "Analog", "IN2P" },
+
+	{ "IN2L Mode", "Digital", "SPKRXDAT" },
+	{ "IN2R Mode", "Digital", "SPKRXDAT" },
+
+	{ "IN2L", NULL, "IN2L Mode" },
+	{ "IN2R", NULL, "IN2R Mode" },
+
+	MADERA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+	MADERA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+	MADERA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+	MADERA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+	MADERA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+
+	MADERA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+	MADERA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+	MADERA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+	MADERA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+	MADERA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+	MADERA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+	MADERA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+	MADERA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+
+	MADERA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+	MADERA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+	MADERA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"),
+	MADERA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"),
+
+	MADERA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+	MADERA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+	MADERA_MUX_ROUTES("SPD1TX1", "SPDIF1TX1"),
+	MADERA_MUX_ROUTES("SPD1TX2", "SPDIF1TX2"),
+
+	MADERA_MIXER_ROUTES("EQ1", "EQ1"),
+	MADERA_MIXER_ROUTES("EQ2", "EQ2"),
+	MADERA_MIXER_ROUTES("EQ3", "EQ3"),
+	MADERA_MIXER_ROUTES("EQ4", "EQ4"),
+
+	MADERA_MIXER_ROUTES("DRC1L", "DRC1L"),
+	MADERA_MIXER_ROUTES("DRC1R", "DRC1R"),
+	MADERA_MIXER_ROUTES("DRC2L", "DRC2L"),
+	MADERA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+	MADERA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	MADERA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	MADERA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	MADERA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	MADERA_DSP_ROUTES("DSP1"),
+
+	{ "DSP Trigger Out", NULL, "DSP1 Trigger Output" },
+
+	{ "DSP1 Trigger Output", "Switch", "DSP1" },
+
+	MADERA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+	MADERA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"),
+	MADERA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"),
+	MADERA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"),
+
+	MADERA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+	MADERA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
+	MADERA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"),
+	MADERA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"),
+
+	MADERA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+	MADERA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+	MADERA_MUX_ROUTES("ISRC2INT3", "ISRC2INT3"),
+	MADERA_MUX_ROUTES("ISRC2INT4", "ISRC2INT4"),
+
+	MADERA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+	MADERA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
+	MADERA_MUX_ROUTES("ISRC2DEC3", "ISRC2DEC3"),
+	MADERA_MUX_ROUTES("ISRC2DEC4", "ISRC2DEC4"),
+
+	{ "AEC1 Loopback", "HPOUT1L", "OUT1L" },
+	{ "AEC1 Loopback", "HPOUT1R", "OUT1R" },
+	{ "AEC2 Loopback", "HPOUT1L", "OUT1L" },
+	{ "AEC2 Loopback", "HPOUT1R", "OUT1R" },
+	{ "HPOUT1 Demux", NULL, "OUT1L" },
+	{ "HPOUT1 Demux", NULL, "OUT1R" },
+	{ "HPOUTL", "HPOUT", "HPOUT1 Demux" },
+	{ "HPOUTR", "HPOUT", "HPOUT1 Demux" },
+	{ "EPOUTP", "EPOUT", "HPOUT1 Demux" },
+	{ "EPOUTN", "EPOUT", "HPOUT1 Demux" },
+
+	{ "AEC1 Loopback", "SPKOUTL", "OUT4L" },
+	{ "AEC2 Loopback", "SPKOUTL", "OUT4L" },
+	{ "SPKOUTN", NULL, "OUT4L" },
+	{ "SPKOUTP", NULL, "OUT4L" },
+
+	{ "AEC1 Loopback", "SPKDAT1L", "OUT5L" },
+	{ "AEC1 Loopback", "SPKDAT1R", "OUT5R" },
+	{ "AEC2 Loopback", "SPKDAT1L", "OUT5L" },
+	{ "AEC2 Loopback", "SPKDAT1R", "OUT5R" },
+	{ "SPKDAT1L", NULL, "OUT5L" },
+	{ "SPKDAT1R", NULL, "OUT5R" },
+
+	{ "SPDIF1", NULL, "SPD1" },
+
+	{ "MICSUPP", NULL, "SYSCLK" },
+
+	{ "DRC1 Signal Activity", NULL, "DRC1 Activity Output" },
+	{ "DRC2 Signal Activity", NULL, "DRC2 Activity Output" },
+	{ "DRC1 Activity Output", "Switch", "DRC1L" },
+	{ "DRC1 Activity Output", "Switch", "DRC1R" },
+	{ "DRC2 Activity Output", "Switch", "DRC2L" },
+	{ "DRC2 Activity Output", "Switch", "DRC2R" },
+};
+
+static int cs47l15_set_fll(struct snd_soc_component *component, int fll_id,
+			   int source, unsigned int fref, unsigned int fout)
+{
+	struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component);
+
+	switch (fll_id) {
+	case MADERA_FLL1_REFCLK:
+		return madera_set_fll_refclk(&cs47l15->fll[0], source, fref,
+					     fout);
+	case MADERA_FLLAO_REFCLK:
+		return madera_set_fll_ao_refclk(&cs47l15->fll[1], source, fref,
+						fout);
+	case MADERA_FLL1_SYNCCLK:
+		return madera_set_fll_syncclk(&cs47l15->fll[0], source, fref,
+					      fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+static struct snd_soc_dai_driver cs47l15_dai[] = {
+	{
+		.name = "cs47l15-aif1",
+		.id = 1,
+		.base = MADERA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF1 Capture",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "cs47l15-aif2",
+		.id = 2,
+		.base = MADERA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF2 Capture",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "cs47l15-aif3",
+		.id = 3,
+		.base = MADERA_AIF3_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF3 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "cs47l15-cpu-trace",
+		.capture = {
+			.stream_name = "Audio Trace CPU",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.compress_new = snd_soc_new_compress,
+	},
+	{
+		.name = "cs47l15-dsp-trace",
+		.capture = {
+			.stream_name = "Audio Trace DSP",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+	},
+};
+
+static int cs47l15_open(struct snd_compr_stream *stream)
+{
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	struct snd_soc_component *component =
+		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+	struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component);
+	struct madera_priv *priv = &cs47l15->core;
+	struct madera *madera = priv->madera;
+	int n_adsp;
+
+	if (strcmp(rtd->codec_dai->name, "cs47l15-dsp-trace") == 0) {
+		n_adsp = 0;
+	} else {
+		dev_err(madera->dev,
+			"No suitable compressed stream for DAI '%s'\n",
+			rtd->codec_dai->name);
+		return -EINVAL;
+	}
+
+	return wm_adsp_compr_open(&priv->adsp[n_adsp], stream);
+}
+
+static irqreturn_t cs47l15_adsp2_irq(int irq, void *data)
+{
+	struct cs47l15 *cs47l15 = data;
+	struct madera_priv *priv = &cs47l15->core;
+	struct madera *madera = priv->madera;
+	int ret;
+
+	ret = wm_adsp_compr_handle_irq(&priv->adsp[0]);
+	if (ret == -ENODEV) {
+		dev_err(madera->dev, "Spurious compressed data IRQ\n");
+		return IRQ_NONE;
+	}
+
+	return IRQ_HANDLED;
+}
+
+static int cs47l15_component_probe(struct snd_soc_component *component)
+{
+	struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component);
+	struct madera *madera = cs47l15->core.madera;
+	int ret;
+
+	snd_soc_component_init_regmap(component, madera->regmap);
+
+	mutex_lock(&madera->dapm_ptr_lock);
+	madera->dapm = snd_soc_component_get_dapm(component);
+	mutex_unlock(&madera->dapm_ptr_lock);
+
+	ret = madera_init_inputs(component);
+	if (ret)
+		return ret;
+
+	ret = madera_init_outputs(component, CS47L15_MONO_OUTPUTS);
+	if (ret)
+		return ret;
+
+	snd_soc_component_disable_pin(component, "HAPTICS");
+
+	ret = snd_soc_add_component_controls(component,
+					     madera_adsp_rate_controls,
+					     CS47L15_NUM_ADSP);
+	if (ret)
+		return ret;
+
+	wm_adsp2_component_probe(&cs47l15->core.adsp[0], component);
+
+	return 0;
+}
+
+static void cs47l15_component_remove(struct snd_soc_component *component)
+{
+	struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component);
+	struct madera *madera = cs47l15->core.madera;
+
+	mutex_lock(&madera->dapm_ptr_lock);
+	madera->dapm = NULL;
+	mutex_unlock(&madera->dapm_ptr_lock);
+
+	wm_adsp2_component_remove(&cs47l15->core.adsp[0], component);
+}
+
+#define CS47L15_DIG_VU 0x0200
+
+static unsigned int cs47l15_digital_vu[] = {
+	MADERA_DAC_DIGITAL_VOLUME_1L,
+	MADERA_DAC_DIGITAL_VOLUME_1R,
+	MADERA_DAC_DIGITAL_VOLUME_4L,
+	MADERA_DAC_DIGITAL_VOLUME_5L,
+	MADERA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static const struct snd_compr_ops cs47l15_compr_ops = {
+	.open = &cs47l15_open,
+	.free = &wm_adsp_compr_free,
+	.set_params = &wm_adsp_compr_set_params,
+	.get_caps = &wm_adsp_compr_get_caps,
+	.trigger = &wm_adsp_compr_trigger,
+	.pointer = &wm_adsp_compr_pointer,
+	.copy = &wm_adsp_compr_copy,
+};
+
+static const struct snd_soc_component_driver soc_component_dev_cs47l15 = {
+	.probe			= &cs47l15_component_probe,
+	.remove			= &cs47l15_component_remove,
+	.set_sysclk		= &madera_set_sysclk,
+	.set_pll		= &cs47l15_set_fll,
+	.name			= DRV_NAME,
+	.compr_ops		= &cs47l15_compr_ops,
+	.controls		= cs47l15_snd_controls,
+	.num_controls		= ARRAY_SIZE(cs47l15_snd_controls),
+	.dapm_widgets		= cs47l15_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(cs47l15_dapm_widgets),
+	.dapm_routes		= cs47l15_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(cs47l15_dapm_routes),
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static int cs47l15_probe(struct platform_device *pdev)
+{
+	struct madera *madera = dev_get_drvdata(pdev->dev.parent);
+	struct cs47l15 *cs47l15;
+	int i, ret;
+
+	BUILD_BUG_ON(ARRAY_SIZE(cs47l15_dai) > MADERA_MAX_DAI);
+
+	/* quick exit if Madera irqchip driver hasn't completed probe */
+	if (!madera->irq_dev) {
+		dev_dbg(&pdev->dev, "irqchip driver not ready\n");
+		return -EPROBE_DEFER;
+	}
+
+	cs47l15 = devm_kzalloc(&pdev->dev, sizeof(struct cs47l15),
+			       GFP_KERNEL);
+	if (!cs47l15)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, cs47l15);
+
+	cs47l15->core.madera = madera;
+	cs47l15->core.dev = &pdev->dev;
+	cs47l15->core.num_inputs = 4;
+
+	ret = madera_core_init(&cs47l15->core);
+	if (ret)
+		return ret;
+
+	ret = madera_init_overheat(&cs47l15->core);
+	if (ret)
+		goto error_core;
+
+	ret = madera_request_irq(madera, MADERA_IRQ_DSP_IRQ1,
+				 "ADSP2 Compressed IRQ", cs47l15_adsp2_irq,
+				 cs47l15);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to request DSP IRQ: %d\n", ret);
+		goto error_overheat;
+	}
+
+	ret = madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 1);
+	if (ret)
+		dev_warn(&pdev->dev, "Failed to set DSP IRQ wake: %d\n", ret);
+
+	cs47l15->core.adsp[0].part = "cs47l15";
+	cs47l15->core.adsp[0].num = 1;
+	cs47l15->core.adsp[0].type = WMFW_ADSP2;
+	cs47l15->core.adsp[0].rev = 2;
+	cs47l15->core.adsp[0].dev = madera->dev;
+	cs47l15->core.adsp[0].regmap = madera->regmap_32bit;
+
+	cs47l15->core.adsp[0].base = MADERA_DSP1_CONFIG_1;
+	cs47l15->core.adsp[0].mem = cs47l15_dsp1_regions;
+	cs47l15->core.adsp[0].num_mems = ARRAY_SIZE(cs47l15_dsp1_regions);
+
+	cs47l15->core.adsp[0].lock_regions =
+		WM_ADSP2_REGION_1 | WM_ADSP2_REGION_2 | WM_ADSP2_REGION_3;
+
+	ret = wm_adsp2_init(&cs47l15->core.adsp[0]);
+	if (ret != 0)
+		goto error_dsp_irq;
+
+	ret = madera_init_bus_error_irq(&cs47l15->core, 0, wm_adsp2_bus_error);
+	if (ret)
+		goto error_adsp;
+
+	madera_init_fll(madera, 1, MADERA_FLL1_CONTROL_1 - 1,
+			&cs47l15->fll[0]);
+	madera_init_fll(madera, 4, MADERA_FLLAO_CONTROL_1 - 1,
+			&cs47l15->fll[1]);
+
+	for (i = 0; i < ARRAY_SIZE(cs47l15_dai); i++)
+		madera_init_dai(&cs47l15->core, i);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(cs47l15_digital_vu); i++)
+		regmap_update_bits(madera->regmap, cs47l15_digital_vu[i],
+				   CS47L15_DIG_VU, CS47L15_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &soc_component_dev_cs47l15,
+					      cs47l15_dai,
+					      ARRAY_SIZE(cs47l15_dai));
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register component: %d\n", ret);
+		goto error_pm_runtime;
+	}
+
+	return ret;
+
+error_pm_runtime:
+	pm_runtime_disable(&pdev->dev);
+	madera_free_bus_error_irq(&cs47l15->core, 0);
+error_adsp:
+	wm_adsp2_remove(&cs47l15->core.adsp[0]);
+error_dsp_irq:
+	madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 0);
+	madera_free_irq(madera, MADERA_IRQ_DSP_IRQ1, cs47l15);
+error_overheat:
+	madera_free_overheat(&cs47l15->core);
+error_core:
+	madera_core_free(&cs47l15->core);
+
+	return ret;
+}
+
+static int cs47l15_remove(struct platform_device *pdev)
+{
+	struct cs47l15 *cs47l15 = platform_get_drvdata(pdev);
+
+	pm_runtime_disable(&pdev->dev);
+
+	madera_free_bus_error_irq(&cs47l15->core, 0);
+
+	wm_adsp2_remove(&cs47l15->core.adsp[0]);
+
+	madera_set_irq_wake(cs47l15->core.madera, MADERA_IRQ_DSP_IRQ1, 0);
+	madera_free_irq(cs47l15->core.madera, MADERA_IRQ_DSP_IRQ1, cs47l15);
+	madera_free_overheat(&cs47l15->core);
+	madera_core_free(&cs47l15->core);
+
+	return 0;
+}
+
+static struct platform_driver cs47l15_codec_driver = {
+	.driver = {
+		.name = "cs47l15-codec",
+	},
+	.probe = &cs47l15_probe,
+	.remove = &cs47l15_remove,
+};
+
+module_platform_driver(cs47l15_codec_driver);
+
+MODULE_SOFTDEP("pre: madera irq-madera arizona-micsupp");
+MODULE_DESCRIPTION("ASoC CS47L15 driver");
+MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.cirrus.com>");
+MODULE_AUTHOR("Jaswinder Jassal <jjassal@opensource.cirrus.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cs47l15-codec");
diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c
index 5f1e32a5a855..1c00e2a115bc 100644
--- a/sound/soc/codecs/madera.c
+++ b/sound/soc/codecs/madera.c
@@ -1041,6 +1041,10 @@ static void madera_configure_input_mode(struct madera *madera)
 	int max_analogue_inputs, max_dmic_sup, i;
 
 	switch (madera->type) {
+	case CS47L15:
+		max_analogue_inputs = 1;
+		max_dmic_sup = 2;
+		break;
 	case CS47L35:
 		max_analogue_inputs = 2;
 		max_dmic_sup = 2;
-- 
cgit v1.2.3-59-g8ed1b


From 6535e831b4c867467d838f2c7866b4dce1d74679 Mon Sep 17 00:00:00 2001
From: Stuart Henderson <stuarth@opensource.wolfsonmicro.com>
Date: Thu, 25 Jul 2019 17:39:31 +0100
Subject: ASoC: cs47l92: Add codec driver for Cirrus Logic CS47L92

Adds the codec driver for the CS47L92 SmartCodec. This is a
multi-functional codec based on the Cirrus Logic Madera platform.

Signed-off-by: Stuart Henderson <stuarth@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190725163931.24964-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig   |    6 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/cs47l92.c | 2039 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/madera.c  |  430 +++++++++-
 sound/soc/codecs/madera.h  |   10 +
 5 files changed, 2478 insertions(+), 9 deletions(-)
 create mode 100644 sound/soc/codecs/cs47l92.c

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index dc23476d2eea..da4c1ae89742 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS47L35 if MFD_CS47L35
 	select SND_SOC_CS47L85 if MFD_CS47L85
 	select SND_SOC_CS47L90 if MFD_CS47L90
+	select SND_SOC_CS47L92 if MFD_CS47L92
 	select SND_SOC_CS53L30 if I2C
 	select SND_SOC_CX20442 if TTY
 	select SND_SOC_CX2072X if I2C
@@ -597,6 +598,9 @@ config SND_SOC_CS47L85
 config SND_SOC_CS47L90
 	tristate
 
+config SND_SOC_CS47L92
+	tristate
+
 # Cirrus Logic Quad-Channel ADC
 config SND_SOC_CS53L30
 	tristate "Cirrus Logic CS53L30 CODEC"
@@ -730,10 +734,12 @@ config SND_SOC_MADERA
 	default y if SND_SOC_CS47L35=y
 	default y if SND_SOC_CS47L85=y
 	default y if SND_SOC_CS47L90=y
+	default y if SND_SOC_CS47L92=y
 	default m if SND_SOC_CS47L15=m
 	default m if SND_SOC_CS47L35=m
 	default m if SND_SOC_CS47L85=m
 	default m if SND_SOC_CS47L90=m
+	default m if SND_SOC_CS47L92=m
 
 config SND_SOC_MAX98088
 	tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 174346b1418a..9230016b0f9f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -69,6 +69,7 @@ snd-soc-cs47l24-objs := cs47l24.o
 snd-soc-cs47l35-objs := cs47l35.o
 snd-soc-cs47l85-objs := cs47l85.o
 snd-soc-cs47l90-objs := cs47l90.o
+snd-soc-cs47l92-objs := cs47l92.o
 snd-soc-cs53l30-objs := cs53l30.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-cx2072x-objs := cx2072x.o
@@ -351,6 +352,7 @@ obj-$(CONFIG_SND_SOC_CS47L15)	+= snd-soc-cs47l15.o
 obj-$(CONFIG_SND_SOC_CS47L35)	+= snd-soc-cs47l35.o
 obj-$(CONFIG_SND_SOC_CS47L85)	+= snd-soc-cs47l85.o
 obj-$(CONFIG_SND_SOC_CS47L90)	+= snd-soc-cs47l90.o
+obj-$(CONFIG_SND_SOC_CS47L92)	+= snd-soc-cs47l92.o
 obj-$(CONFIG_SND_SOC_CS53L30)	+= snd-soc-cs53l30.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_CX2072X)	+= snd-soc-cx2072x.o
diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c
new file mode 100644
index 000000000000..d50f75f3b3e4
--- /dev/null
+++ b/sound/soc/codecs/cs47l92.c
@@ -0,0 +1,2039 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// ALSA SoC Audio driver for CS47L92 codec
+//
+// Copyright (C) 2016-2019 Cirrus Logic, Inc. and
+//                         Cirrus Logic International Semiconductor Ltd.
+//
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include <linux/irqchip/irq-madera.h>
+#include <linux/mfd/madera/core.h>
+#include <linux/mfd/madera/registers.h>
+
+#include "madera.h"
+#include "wm_adsp.h"
+
+#define CS47L92_NUM_ADSP	1
+#define CS47L92_MONO_OUTPUTS	3
+
+#define DRV_NAME "cs47l92-codec"
+
+struct cs47l92 {
+	struct madera_priv core;
+	struct madera_fll fll[2];
+};
+
+static const struct wm_adsp_region cs47l92_dsp1_regions[] = {
+	{ .type = WMFW_ADSP2_PM, .base = 0x080000 },
+	{ .type = WMFW_ADSP2_ZM, .base = 0x0e0000 },
+	{ .type = WMFW_ADSP2_XM, .base = 0x0a0000 },
+	{ .type = WMFW_ADSP2_YM, .base = 0x0c0000 },
+};
+
+static const char * const cs47l92_outdemux_texts[] = {
+	"HPOUT3",
+	"HPOUT4",
+};
+
+static int cs47l92_put_demux(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_kcontrol_component(kcontrol);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	struct cs47l92 *cs47l92 = snd_soc_component_get_drvdata(component);
+	struct madera_priv *priv = &cs47l92->core;
+	struct madera *madera = priv->madera;
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int ep_sel, mux, change, cur;
+	bool out_mono;
+	int ret;
+
+	if (ucontrol->value.enumerated.item[0] > e->items - 1)
+		return -EINVAL;
+
+	mux = ucontrol->value.enumerated.item[0];
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	ep_sel = mux << e->shift_l;
+
+	change = snd_soc_component_test_bits(component, MADERA_OUTPUT_ENABLES_1,
+					     MADERA_EP_SEL_MASK,
+					     ep_sel);
+	if (!change)
+		goto end;
+
+	ret = regmap_read(madera->regmap, MADERA_OUTPUT_ENABLES_1, &cur);
+	if (ret != 0)
+		dev_warn(madera->dev, "Failed to read outputs: %d\n", ret);
+
+	/* EP_SEL should not be modified while HPOUT3 or 4 is enabled */
+	ret = regmap_update_bits(madera->regmap, MADERA_OUTPUT_ENABLES_1,
+				 MADERA_OUT3L_ENA | MADERA_OUT3R_ENA, 0);
+	if (ret)
+		dev_warn(madera->dev, "Failed to disable outputs: %d\n", ret);
+
+	usleep_range(2000, 3000); /* wait for wseq to complete */
+
+	ret = regmap_update_bits(madera->regmap, MADERA_OUTPUT_ENABLES_1,
+				 MADERA_EP_SEL, ep_sel);
+	if (ret) {
+		dev_err(madera->dev, "Failed to set OUT3 demux: %d\n", ret);
+	} else {
+		out_mono = madera->pdata.codec.out_mono[2 + mux];
+
+		ret = madera_set_output_mode(component, 3, out_mono);
+		if (ret < 0)
+			dev_warn(madera->dev,
+				 "Failed to set output mode: %d\n", ret);
+	}
+
+	ret = regmap_update_bits(madera->regmap, MADERA_OUTPUT_ENABLES_1,
+				 MADERA_OUT3L_ENA | MADERA_OUT3R_ENA, cur);
+	if (ret) {
+		dev_warn(madera->dev, "Failed to restore outputs: %d\n", ret);
+	} else {
+		/* wait for wseq */
+		if (cur & (MADERA_OUT3L_ENA | MADERA_OUT3R_ENA))
+			msleep(34); /* enable delay */
+		else
+			usleep_range(2000, 3000); /* disable delay */
+	}
+
+end:
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+}
+
+static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum,
+			    MADERA_OUTPUT_ENABLES_1,
+			    MADERA_EP_SEL_SHIFT,
+			    cs47l92_outdemux_texts);
+
+static const struct snd_kcontrol_new cs47l92_outdemux =
+	SOC_DAPM_ENUM_EXT("OUT3 Demux", cs47l92_outdemux_enum,
+			  snd_soc_dapm_get_enum_double, cs47l92_put_demux);
+
+static int cs47l92_adsp_power_ev(struct snd_soc_dapm_widget *w,
+				 struct snd_kcontrol *kcontrol,
+				 int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct cs47l92 *cs47l92 = snd_soc_component_get_drvdata(component);
+	struct madera_priv *priv = &cs47l92->core;
+	struct madera *madera = priv->madera;
+	unsigned int freq;
+	int ret;
+
+	ret = regmap_read(madera->regmap, MADERA_DSP_CLOCK_2, &freq);
+	if (ret != 0) {
+		dev_err(madera->dev,
+			"Failed to read MADERA_DSP_CLOCK_2: %d\n", ret);
+		return ret;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		ret = madera_set_adsp_clk(&cs47l92->core, w->shift, freq);
+		if (ret)
+			return ret;
+		break;
+	default:
+		break;
+	}
+
+	return wm_adsp_early_event(w, kcontrol, event);
+}
+
+#define CS47L92_NG_SRC(name, base) \
+	SOC_SINGLE(name " NG HPOUT1L Switch",  base,  0, 1, 0), \
+	SOC_SINGLE(name " NG HPOUT1R Switch",  base,  1, 1, 0), \
+	SOC_SINGLE(name " NG HPOUT2L Switch",  base,  2, 1, 0), \
+	SOC_SINGLE(name " NG HPOUT2R Switch",  base,  3, 1, 0), \
+	SOC_SINGLE(name " NG HPOUT3L Switch",  base,  4, 1, 0), \
+	SOC_SINGLE(name " NG HPOUT3R Switch",  base,  5, 1, 0), \
+	SOC_SINGLE(name " NG SPKDAT1L Switch", base,  8, 1, 0), \
+	SOC_SINGLE(name " NG SPKDAT1R Switch", base,  9, 1, 0)
+
+static const struct snd_kcontrol_new cs47l92_snd_controls[] = {
+SOC_ENUM("IN1 OSR", madera_in_dmic_osr[0]),
+SOC_ENUM("IN2 OSR", madera_in_dmic_osr[1]),
+SOC_ENUM("IN3 OSR", madera_in_dmic_osr[2]),
+SOC_ENUM("IN4 OSR", madera_in_dmic_osr[3]),
+
+SOC_SINGLE_RANGE_TLV("IN1L Volume", MADERA_IN1L_CONTROL,
+		     MADERA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN1R Volume", MADERA_IN1R_CONTROL,
+		     MADERA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN2L Volume", MADERA_IN2L_CONTROL,
+		     MADERA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN2R Volume", MADERA_IN2R_CONTROL,
+		     MADERA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv),
+
+SOC_ENUM("IN HPF Cutoff Frequency", madera_in_hpf_cut_enum),
+
+SOC_SINGLE_EXT("IN1L LP Switch", MADERA_ADC_DIGITAL_VOLUME_1L,
+	       MADERA_IN1L_LP_MODE_SHIFT, 1, 0,
+	       snd_soc_get_volsw, madera_lp_mode_put),
+SOC_SINGLE_EXT("IN1R LP Switch", MADERA_ADC_DIGITAL_VOLUME_1R,
+	       MADERA_IN1L_LP_MODE_SHIFT, 1, 0,
+	       snd_soc_get_volsw, madera_lp_mode_put),
+SOC_SINGLE_EXT("IN2L LP Switch", MADERA_ADC_DIGITAL_VOLUME_2L,
+	       MADERA_IN1L_LP_MODE_SHIFT, 1, 0,
+	       snd_soc_get_volsw, madera_lp_mode_put),
+SOC_SINGLE_EXT("IN2R LP Switch", MADERA_ADC_DIGITAL_VOLUME_2R,
+	       MADERA_IN1L_LP_MODE_SHIFT, 1, 0,
+	       snd_soc_get_volsw, madera_lp_mode_put),
+
+SOC_SINGLE("IN1L HPF Switch", MADERA_IN1L_CONTROL,
+	   MADERA_IN1L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN1R HPF Switch", MADERA_IN1R_CONTROL,
+	   MADERA_IN1R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2L HPF Switch", MADERA_IN2L_CONTROL,
+	   MADERA_IN2L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2R HPF Switch", MADERA_IN2R_CONTROL,
+	   MADERA_IN2R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN3L HPF Switch", MADERA_IN3L_CONTROL,
+	   MADERA_IN3L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN3R HPF Switch", MADERA_IN3R_CONTROL,
+	   MADERA_IN3R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN4L HPF Switch", MADERA_IN4L_CONTROL,
+	   MADERA_IN4L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN4R HPF Switch", MADERA_IN4R_CONTROL,
+	   MADERA_IN4R_HPF_SHIFT, 1, 0),
+
+SOC_SINGLE_TLV("IN1L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1L,
+	       MADERA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN1R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1R,
+	       MADERA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN2L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2L,
+	       MADERA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN2R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2R,
+	       MADERA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN3L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_3L,
+	       MADERA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN3R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_3R,
+	       MADERA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN4L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_4L,
+	       MADERA_IN4L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+SOC_SINGLE_TLV("IN4R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_4R,
+	       MADERA_IN4R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv),
+
+SOC_ENUM("Input Ramp Up", madera_in_vi_ramp),
+SOC_ENUM("Input Ramp Down", madera_in_vd_ramp),
+
+MADERA_MIXER_CONTROLS("EQ1", MADERA_EQ1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("EQ2", MADERA_EQ2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("EQ3", MADERA_EQ3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("EQ4", MADERA_EQ4MIX_INPUT_1_SOURCE),
+
+MADERA_EQ_CONTROL("EQ1 Coefficients", MADERA_EQ1_2),
+SOC_SINGLE_TLV("EQ1 B1 Volume", MADERA_EQ1_1, MADERA_EQ1_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", MADERA_EQ1_1, MADERA_EQ1_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", MADERA_EQ1_1, MADERA_EQ1_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", MADERA_EQ1_2, MADERA_EQ1_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", MADERA_EQ1_2, MADERA_EQ1_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+MADERA_EQ_CONTROL("EQ2 Coefficients", MADERA_EQ2_2),
+SOC_SINGLE_TLV("EQ2 B1 Volume", MADERA_EQ2_1, MADERA_EQ2_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", MADERA_EQ2_1, MADERA_EQ2_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", MADERA_EQ2_1, MADERA_EQ2_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", MADERA_EQ2_2, MADERA_EQ2_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", MADERA_EQ2_2, MADERA_EQ2_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+MADERA_EQ_CONTROL("EQ3 Coefficients", MADERA_EQ3_2),
+SOC_SINGLE_TLV("EQ3 B1 Volume", MADERA_EQ3_1, MADERA_EQ3_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", MADERA_EQ3_1, MADERA_EQ3_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", MADERA_EQ3_1, MADERA_EQ3_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", MADERA_EQ3_2, MADERA_EQ3_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", MADERA_EQ3_2, MADERA_EQ3_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+MADERA_EQ_CONTROL("EQ4 Coefficients", MADERA_EQ4_2),
+SOC_SINGLE_TLV("EQ4 B1 Volume", MADERA_EQ4_1, MADERA_EQ4_B1_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", MADERA_EQ4_1, MADERA_EQ4_B2_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", MADERA_EQ4_1, MADERA_EQ4_B3_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", MADERA_EQ4_2, MADERA_EQ4_B4_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", MADERA_EQ4_2, MADERA_EQ4_B5_GAIN_SHIFT,
+	       24, 0, madera_eq_tlv),
+
+SOC_SINGLE("DAC High Performance Mode Switch", MADERA_OUTPUT_RATE_1,
+	   MADERA_CP_DAC_MODE_SHIFT, 1, 0),
+
+MADERA_MIXER_CONTROLS("DRC1L", MADERA_DRC1LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DRC1R", MADERA_DRC1RMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DRC2L", MADERA_DRC2LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DRC2R", MADERA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", MADERA_DRC1_CTRL1, 5,
+		   MADERA_DRC1R_ENA | MADERA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", MADERA_DRC2_CTRL1, 5,
+		   MADERA_DRC2R_ENA | MADERA_DRC2L_ENA),
+
+MADERA_MIXER_CONTROLS("LHPF1", MADERA_HPLP1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("LHPF2", MADERA_HPLP2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("LHPF3", MADERA_HPLP3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("LHPF4", MADERA_HPLP4MIX_INPUT_1_SOURCE),
+
+MADERA_LHPF_CONTROL("LHPF1 Coefficients", MADERA_HPLPF1_2),
+MADERA_LHPF_CONTROL("LHPF2 Coefficients", MADERA_HPLPF2_2),
+MADERA_LHPF_CONTROL("LHPF3 Coefficients", MADERA_HPLPF3_2),
+MADERA_LHPF_CONTROL("LHPF4 Coefficients", MADERA_HPLPF4_2),
+
+SOC_ENUM("LHPF1 Mode", madera_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", madera_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", madera_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", madera_lhpf4_mode),
+
+MADERA_RATE_ENUM("ISRC1 FSL", madera_isrc_fsl[0]),
+MADERA_RATE_ENUM("ISRC2 FSL", madera_isrc_fsl[1]),
+MADERA_RATE_ENUM("ISRC1 FSH", madera_isrc_fsh[0]),
+MADERA_RATE_ENUM("ISRC2 FSH", madera_isrc_fsh[1]),
+MADERA_RATE_ENUM("ASRC1 Rate 1", madera_asrc1_bidir_rate[0]),
+MADERA_RATE_ENUM("ASRC1 Rate 2", madera_asrc1_bidir_rate[1]),
+
+WM_ADSP2_PRELOAD_SWITCH("DSP1", 1),
+
+MADERA_MIXER_CONTROLS("DSP1L", MADERA_DSP1LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("DSP1R", MADERA_DSP1RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", MADERA_COMFORT_NOISE_GENERATOR,
+	       MADERA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, madera_noise_tlv),
+
+MADERA_MIXER_CONTROLS("HPOUT1L", MADERA_OUT1LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("HPOUT1R", MADERA_OUT1RMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("HPOUT2L", MADERA_OUT2LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("HPOUT2R", MADERA_OUT2RMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("HPOUT3L", MADERA_OUT3LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("HPOUT3R", MADERA_OUT3RMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SPKDAT1L", MADERA_OUT5LMIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SPKDAT1R", MADERA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 SC Protect Switch", MADERA_HP1_SHORT_CIRCUIT_CTRL,
+	   MADERA_HP1_SC_ENA_SHIFT, 1, 0),
+SOC_SINGLE("HPOUT2 SC Protect Switch", MADERA_HP2_SHORT_CIRCUIT_CTRL,
+	   MADERA_HP2_SC_ENA_SHIFT, 1, 0),
+SOC_SINGLE("HPOUT3 SC Protect Switch", MADERA_HP3_SHORT_CIRCUIT_CTRL,
+	   MADERA_HP3_SC_ENA_SHIFT, 1, 0),
+
+SOC_SINGLE("SPKDAT1 High Performance Switch", MADERA_OUTPUT_PATH_CONFIG_5L,
+	   MADERA_OUT5_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_1L,
+	     MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("HPOUT2 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_2L,
+	     MADERA_DAC_DIGITAL_VOLUME_2R, MADERA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("HPOUT3 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_3L,
+	     MADERA_DAC_DIGITAL_VOLUME_3R, MADERA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_5L,
+	     MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_1L,
+		 MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, madera_digital_tlv),
+SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_2L,
+		 MADERA_DAC_DIGITAL_VOLUME_2R, MADERA_OUT2L_VOL_SHIFT,
+		 0xbf, 0, madera_digital_tlv),
+SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_3L,
+		 MADERA_DAC_DIGITAL_VOLUME_3R, MADERA_OUT3L_VOL_SHIFT,
+		 0xbf, 0, madera_digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_5L,
+		 MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, madera_digital_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", MADERA_PDM_SPK1_CTRL_1, MADERA_SPK1L_MUTE_SHIFT,
+	   MADERA_SPK1R_MUTE_SHIFT, 1, 1),
+
+SOC_ENUM("Output Ramp Up", madera_out_vi_ramp),
+SOC_ENUM("Output Ramp Down", madera_out_vd_ramp),
+
+SOC_SINGLE("Noise Gate Switch", MADERA_NOISE_GATE_CONTROL,
+	   MADERA_NGATE_ENA_SHIFT, 1, 0),
+SOC_SINGLE_TLV("Noise Gate Threshold Volume", MADERA_NOISE_GATE_CONTROL,
+	       MADERA_NGATE_THR_SHIFT, 7, 1, madera_ng_tlv),
+SOC_ENUM("Noise Gate Hold", madera_ng_hold),
+
+SOC_ENUM_EXT("DFC1RX Width", madera_dfc_width[0],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC1RX Type", madera_dfc_type[0],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC1TX Width", madera_dfc_width[1],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC1TX Type", madera_dfc_type[1],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC2RX Width", madera_dfc_width[2],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC2RX Type", madera_dfc_type[2],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC2TX Width", madera_dfc_width[3],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC2TX Type", madera_dfc_type[3],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC3RX Width", madera_dfc_width[4],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC3RX Type", madera_dfc_type[4],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC3TX Width", madera_dfc_width[5],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC3TX Type", madera_dfc_type[5],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC4RX Width", madera_dfc_width[6],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC4RX Type", madera_dfc_type[6],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC4TX Width", madera_dfc_width[7],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC4TX Type", madera_dfc_type[7],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC5RX Width", madera_dfc_width[8],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC5RX Type", madera_dfc_type[8],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC5TX Width", madera_dfc_width[9],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC5TX Type", madera_dfc_type[9],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC6RX Width", madera_dfc_width[10],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC6RX Type", madera_dfc_type[10],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC6TX Width", madera_dfc_width[11],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC6TX Type", madera_dfc_type[11],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC7RX Width", madera_dfc_width[12],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC7RX Type", madera_dfc_type[12],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC7TX Width", madera_dfc_width[13],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC7TX Type", madera_dfc_type[13],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC8RX Width", madera_dfc_width[14],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC8RX Type", madera_dfc_type[14],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC8TX Width", madera_dfc_width[15],
+	     snd_soc_get_enum_double, madera_dfc_put),
+SOC_ENUM_EXT("DFC8TX Type", madera_dfc_type[15],
+	     snd_soc_get_enum_double, madera_dfc_put),
+
+CS47L92_NG_SRC("HPOUT1L", MADERA_NOISE_GATE_SELECT_1L),
+CS47L92_NG_SRC("HPOUT1R", MADERA_NOISE_GATE_SELECT_1R),
+CS47L92_NG_SRC("HPOUT2L", MADERA_NOISE_GATE_SELECT_2L),
+CS47L92_NG_SRC("HPOUT2R", MADERA_NOISE_GATE_SELECT_2R),
+CS47L92_NG_SRC("HPOUT3L", MADERA_NOISE_GATE_SELECT_3L),
+CS47L92_NG_SRC("HPOUT3R", MADERA_NOISE_GATE_SELECT_3R),
+CS47L92_NG_SRC("SPKDAT1L", MADERA_NOISE_GATE_SELECT_5L),
+CS47L92_NG_SRC("SPKDAT1R", MADERA_NOISE_GATE_SELECT_5R),
+
+MADERA_MIXER_CONTROLS("AIF1TX1", MADERA_AIF1TX1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX2", MADERA_AIF1TX2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX3", MADERA_AIF1TX3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX4", MADERA_AIF1TX4MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX5", MADERA_AIF1TX5MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX6", MADERA_AIF1TX6MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX7", MADERA_AIF1TX7MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF1TX8", MADERA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+MADERA_MIXER_CONTROLS("AIF2TX1", MADERA_AIF2TX1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX2", MADERA_AIF2TX2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX3", MADERA_AIF2TX3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX4", MADERA_AIF2TX4MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX5", MADERA_AIF2TX5MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX6", MADERA_AIF2TX6MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX7", MADERA_AIF2TX7MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF2TX8", MADERA_AIF2TX8MIX_INPUT_1_SOURCE),
+
+MADERA_MIXER_CONTROLS("AIF3TX1", MADERA_AIF3TX1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF3TX2", MADERA_AIF3TX2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF3TX3", MADERA_AIF3TX3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("AIF3TX4", MADERA_AIF3TX4MIX_INPUT_1_SOURCE),
+
+MADERA_MIXER_CONTROLS("SLIMTX1", MADERA_SLIMTX1MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SLIMTX2", MADERA_SLIMTX2MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SLIMTX3", MADERA_SLIMTX3MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SLIMTX4", MADERA_SLIMTX4MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SLIMTX5", MADERA_SLIMTX5MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SLIMTX6", MADERA_SLIMTX6MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SLIMTX7", MADERA_SLIMTX7MIX_INPUT_1_SOURCE),
+MADERA_MIXER_CONTROLS("SLIMTX8", MADERA_SLIMTX8MIX_INPUT_1_SOURCE),
+
+MADERA_GAINMUX_CONTROLS("SPDIFTX1", MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE),
+MADERA_GAINMUX_CONTROLS("SPDIFTX2", MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE),
+
+WM_ADSP_FW_CONTROL("DSP1", 0),
+};
+
+MADERA_MIXER_ENUMS(EQ1, MADERA_EQ1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(EQ2, MADERA_EQ2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(EQ3, MADERA_EQ3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(EQ4, MADERA_EQ4MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(DRC1L, MADERA_DRC1LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DRC1R, MADERA_DRC1RMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DRC2L, MADERA_DRC2LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DRC2R, MADERA_DRC2RMIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(LHPF1, MADERA_HPLP1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(LHPF2, MADERA_HPLP2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(LHPF3, MADERA_HPLP3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(LHPF4, MADERA_HPLP4MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(DSP1L, MADERA_DSP1LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(DSP1R, MADERA_DSP1RMIX_INPUT_1_SOURCE);
+MADERA_DSP_AUX_ENUMS(DSP1, MADERA_DSP1AUX1MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(PWM1, MADERA_PWM1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(PWM2, MADERA_PWM2MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(OUT1L, MADERA_OUT1LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(OUT1R, MADERA_OUT1RMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(OUT2L, MADERA_OUT2LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(OUT2R, MADERA_OUT2RMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(OUT3L, MADERA_OUT3LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(OUT3R, MADERA_OUT3RMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SPKDAT1L, MADERA_OUT5LMIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SPKDAT1R, MADERA_OUT5RMIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(AIF1TX1, MADERA_AIF1TX1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX2, MADERA_AIF1TX2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX3, MADERA_AIF1TX3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX4, MADERA_AIF1TX4MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX5, MADERA_AIF1TX5MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX6, MADERA_AIF1TX6MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX7, MADERA_AIF1TX7MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF1TX8, MADERA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(AIF2TX1, MADERA_AIF2TX1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX2, MADERA_AIF2TX2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX3, MADERA_AIF2TX3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX4, MADERA_AIF2TX4MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX5, MADERA_AIF2TX5MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX6, MADERA_AIF2TX6MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX7, MADERA_AIF2TX7MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF2TX8, MADERA_AIF2TX8MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(AIF3TX1, MADERA_AIF3TX1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF3TX2, MADERA_AIF3TX2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF3TX3, MADERA_AIF3TX3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(AIF3TX4, MADERA_AIF3TX4MIX_INPUT_1_SOURCE);
+
+MADERA_MIXER_ENUMS(SLIMTX1, MADERA_SLIMTX1MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SLIMTX2, MADERA_SLIMTX2MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SLIMTX3, MADERA_SLIMTX3MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SLIMTX4, MADERA_SLIMTX4MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SLIMTX5, MADERA_SLIMTX5MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SLIMTX6, MADERA_SLIMTX6MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SLIMTX7, MADERA_SLIMTX7MIX_INPUT_1_SOURCE);
+MADERA_MIXER_ENUMS(SLIMTX8, MADERA_SLIMTX8MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(SPD1TX1, MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(SPD1TX2, MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ASRC1IN1L, MADERA_ASRC1_1LMIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ASRC1IN1R, MADERA_ASRC1_1RMIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ASRC1IN2L, MADERA_ASRC1_2LMIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ASRC1IN2R, MADERA_ASRC1_2RMIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC1INT1, MADERA_ISRC1INT1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1INT2, MADERA_ISRC1INT2MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC1DEC1, MADERA_ISRC1DEC1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC1DEC2, MADERA_ISRC1DEC2MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC2INT1, MADERA_ISRC2INT1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2INT2, MADERA_ISRC2INT2MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(ISRC2DEC1, MADERA_ISRC2DEC1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(ISRC2DEC2, MADERA_ISRC2DEC2MIX_INPUT_1_SOURCE);
+
+MADERA_MUX_ENUMS(DFC1, MADERA_DFC1MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(DFC2, MADERA_DFC2MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(DFC3, MADERA_DFC3MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(DFC4, MADERA_DFC4MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(DFC5, MADERA_DFC5MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(DFC6, MADERA_DFC6MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(DFC7, MADERA_DFC7MIX_INPUT_1_SOURCE);
+MADERA_MUX_ENUMS(DFC8, MADERA_DFC8MIX_INPUT_1_SOURCE);
+
+static const char * const cs47l92_aec_loopback_texts[] = {
+	"HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "HPOUT3L", "HPOUT3R",
+	"SPKDAT1L", "SPKDAT1R",
+};
+
+static const unsigned int cs47l92_aec_loopback_values[] = {
+	0, 1, 2, 3, 4, 5, 8, 9
+};
+
+static const struct soc_enum cs47l92_aec_loopback =
+	SOC_VALUE_ENUM_SINGLE(MADERA_DAC_AEC_CONTROL_1,
+			      MADERA_AEC1_LOOPBACK_SRC_SHIFT, 0xf,
+			      ARRAY_SIZE(cs47l92_aec_loopback_texts),
+			      cs47l92_aec_loopback_texts,
+			      cs47l92_aec_loopback_values);
+
+static const struct snd_kcontrol_new cs47l92_aec_loopback_mux =
+	SOC_DAPM_ENUM("AEC1 Loopback", cs47l92_aec_loopback);
+
+static const struct snd_soc_dapm_widget cs47l92_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT,
+		    0, madera_sysclk_ev,
+		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", MADERA_ASYNC_CLOCK_1,
+		    MADERA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK,
+		    MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", MADERA_OUTPUT_ASYNC_CLOCK,
+		    MADERA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1,
+		    MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD1", 20, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD2", 20, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", MADERA_MIC_BIAS_CTRL_1,
+		    MADERA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", MADERA_MIC_BIAS_CTRL_2,
+		    MADERA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1A", MADERA_MIC_BIAS_CTRL_5,
+		    MADERA_MICB1A_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS1B", MADERA_MIC_BIAS_CTRL_5,
+		    MADERA_MICB1B_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS1C", MADERA_MIC_BIAS_CTRL_5,
+		    MADERA_MICB1C_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS1D", MADERA_MIC_BIAS_CTRL_5,
+		    MADERA_MICB1D_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS2A", MADERA_MIC_BIAS_CTRL_6,
+		    MADERA_MICB2A_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2B", MADERA_MIC_BIAS_CTRL_6,
+		    MADERA_MICB2B_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("FXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_FX, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("ASRC1CLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_ASRC1, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("ISRC1CLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_ISRC1, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("ISRC2CLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_ISRC2, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("OUTCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_OUT, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("SPDCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_SPD, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_DSP1, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF1TXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_AIF1, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF2TXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_AIF2, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF3TXCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_AIF3, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("SLIMBUSCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_SLIMBUS, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("PWMCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_PWM, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("DFCCLK", SND_SOC_NOPM,
+		    MADERA_DOM_GRP_DFC, 0,
+		    madera_domain_clk_ev,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1ALN"),
+SND_SOC_DAPM_INPUT("IN1ALP"),
+SND_SOC_DAPM_INPUT("IN1BLN"),
+SND_SOC_DAPM_INPUT("IN1BLP"),
+SND_SOC_DAPM_INPUT("IN1ARN"),
+SND_SOC_DAPM_INPUT("IN1ARP"),
+SND_SOC_DAPM_INPUT("IN1BR"),
+SND_SOC_DAPM_INPUT("IN2ALN"),
+SND_SOC_DAPM_INPUT("IN2ALP"),
+SND_SOC_DAPM_INPUT("IN2BL"),
+SND_SOC_DAPM_INPUT("IN2ARN"),
+SND_SOC_DAPM_INPUT("IN2ARP"),
+SND_SOC_DAPM_INPUT("IN2BR"),
+
+SND_SOC_DAPM_MUX("IN1L Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[0]),
+SND_SOC_DAPM_MUX("IN1R Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[1]),
+SND_SOC_DAPM_MUX("IN2L Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[2]),
+SND_SOC_DAPM_MUX("IN2R Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[3]),
+
+SND_SOC_DAPM_MUX("IN1L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]),
+SND_SOC_DAPM_MUX("IN1R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]),
+
+SND_SOC_DAPM_MUX("IN2L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]),
+SND_SOC_DAPM_MUX("IN2R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]),
+
+SND_SOC_DAPM_DEMUX("OUT3 Demux", SND_SOC_NOPM, 0, 0, &cs47l92_outdemux),
+
+SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
+SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM1_ENA_SHIFT,
+		 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM2_ENA_SHIFT,
+		 0, NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+		     MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX7", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX8", NULL, 0,
+		     MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0,
+		     MADERA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     MADERA_SLIMTX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+		     MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+		     MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX3", NULL, 0,
+		     MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX4", NULL, 0,
+		     MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX4_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
+		   MADERA_OUT1L_ENA_SHIFT, 0, NULL, 0, madera_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
+		   MADERA_OUT1R_ENA_SHIFT, 0, NULL, 0, madera_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", SND_SOC_NOPM,
+		   MADERA_OUT2L_ENA_SHIFT, 0, NULL, 0, madera_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", SND_SOC_NOPM,
+		   MADERA_OUT2R_ENA_SHIFT, 0, NULL, 0, madera_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", MADERA_OUTPUT_ENABLES_1,
+		   MADERA_OUT3L_ENA_SHIFT, 0, NULL, 0, madera_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3R", MADERA_OUTPUT_ENABLES_1,
+		   MADERA_OUT3R_ENA_SHIFT, 0, NULL, 0, madera_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", MADERA_OUTPUT_ENABLES_1,
+		   MADERA_OUT5L_ENA_SHIFT, 0, NULL, 0, madera_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", MADERA_OUTPUT_ENABLES_1,
+		   MADERA_OUT5R_ENA_SHIFT, 0, NULL, 0, madera_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_PGA("SPD1TX1", MADERA_SPD1_TX_CONTROL,
+		 MADERA_SPD1_VAL1_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SPD1TX2", MADERA_SPD1_TX_CONTROL,
+		 MADERA_SPD1_VAL2_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("SPD1", MADERA_SPD1_TX_CONTROL,
+		     MADERA_SPD1_ENA_SHIFT, 0, NULL, 0),
+
+/*
+ * mux_in widgets : arranged in the order of sources
+ * specified in MADERA_MIXER_INPUT_ROUTES
+ */
+
+SND_SOC_DAPM_PGA("Noise Generator", MADERA_COMFORT_NOISE_GENERATOR,
+		 MADERA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", MADERA_TONE_GENERATOR_1,
+		 MADERA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", MADERA_TONE_GENERATOR_1,
+		 MADERA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_SIGGEN("HAPTICS"),
+
+SND_SOC_DAPM_MUX("AEC1 Loopback", MADERA_DAC_AEC_CONTROL_1,
+		 MADERA_AEC1_LOOPBACK_ENA_SHIFT, 0,
+		 &cs47l92_aec_loopback_mux),
+
+SND_SOC_DAPM_PGA_E("IN1L", MADERA_INPUT_ENABLES, MADERA_IN1L_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R", MADERA_INPUT_ENABLES, MADERA_IN1R_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L", MADERA_INPUT_ENABLES, MADERA_IN2L_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R", MADERA_INPUT_ENABLES, MADERA_IN2R_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3L", MADERA_INPUT_ENABLES, MADERA_IN3L_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3R", MADERA_INPUT_ENABLES, MADERA_IN3R_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4L", MADERA_INPUT_ENABLES, MADERA_IN4L_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4R", MADERA_INPUT_ENABLES, MADERA_IN4R_ENA_SHIFT,
+		   0, NULL, 0, madera_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+		    MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX7", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX8", NULL, 0,
+		    MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+		    MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+		    MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX3", NULL, 0,
+		    MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX4", NULL, 0,
+		    MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX4_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, MADERA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    MADERA_SLIMRX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA("EQ1", MADERA_EQ1_1, MADERA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", MADERA_EQ2_1, MADERA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", MADERA_EQ3_1, MADERA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", MADERA_EQ4_1, MADERA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", MADERA_DRC1_CTRL1, MADERA_DRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", MADERA_DRC1_CTRL1, MADERA_DRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", MADERA_DRC2_CTRL1, MADERA_DRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", MADERA_DRC2_CTRL1, MADERA_DRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", MADERA_HPLPF1_1, MADERA_LHPF1_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", MADERA_HPLPF2_1, MADERA_LHPF2_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", MADERA_HPLPF3_1, MADERA_LHPF3_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", MADERA_HPLPF4_1, MADERA_LHPF4_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("ASRC1IN1L", MADERA_ASRC1_ENABLE,
+		 MADERA_ASRC1_IN1L_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1IN1R", MADERA_ASRC1_ENABLE,
+		 MADERA_ASRC1_IN1R_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1IN2L", MADERA_ASRC1_ENABLE,
+		 MADERA_ASRC1_IN2L_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1IN2R", MADERA_ASRC1_ENABLE,
+		 MADERA_ASRC1_IN2R_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1DEC1", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC2", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1INT1", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT2", MADERA_ISRC_1_CTRL_3,
+		 MADERA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2DEC1", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC2", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2INT1", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT2", MADERA_ISRC_2_CTRL_3,
+		 MADERA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0),
+
+WM_ADSP2("DSP1", 0, cs47l92_adsp_power_ev),
+
+/* end of ordered widget list */
+
+SND_SOC_DAPM_PGA("DFC1", MADERA_DFC1_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("DFC2", MADERA_DFC2_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("DFC3", MADERA_DFC3_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("DFC4", MADERA_DFC4_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("DFC5", MADERA_DFC5_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("DFC6", MADERA_DFC6_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("DFC7", MADERA_DFC7_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("DFC8", MADERA_DFC8_CTRL, MADERA_DFC1_ENA_SHIFT, 0, NULL, 0),
+
+MADERA_MIXER_WIDGETS(EQ1, "EQ1"),
+MADERA_MIXER_WIDGETS(EQ2, "EQ2"),
+MADERA_MIXER_WIDGETS(EQ3, "EQ3"),
+MADERA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+MADERA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+MADERA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+MADERA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+MADERA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+SND_SOC_DAPM_SWITCH("DRC1 Activity Output", SND_SOC_NOPM, 0, 0,
+		    &madera_drc_activity_output_mux[0]),
+SND_SOC_DAPM_SWITCH("DRC2 Activity Output", SND_SOC_NOPM, 0, 0,
+		    &madera_drc_activity_output_mux[1]),
+
+MADERA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+MADERA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+MADERA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+MADERA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+MADERA_MIXER_WIDGETS(PWM1, "PWM1"),
+MADERA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+MADERA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+MADERA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+MADERA_MIXER_WIDGETS(OUT2L, "HPOUT2L"),
+MADERA_MIXER_WIDGETS(OUT2R, "HPOUT2R"),
+MADERA_MIXER_WIDGETS(OUT3L, "HPOUT3L"),
+MADERA_MIXER_WIDGETS(OUT3R, "HPOUT3R"),
+MADERA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+MADERA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+
+MADERA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+MADERA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+MADERA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+MADERA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+MADERA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+MADERA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+MADERA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+MADERA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+MADERA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+MADERA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+MADERA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"),
+MADERA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"),
+MADERA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"),
+MADERA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"),
+MADERA_MIXER_WIDGETS(AIF2TX7, "AIF2TX7"),
+MADERA_MIXER_WIDGETS(AIF2TX8, "AIF2TX8"),
+
+MADERA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+MADERA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+MADERA_MIXER_WIDGETS(AIF3TX3, "AIF3TX3"),
+MADERA_MIXER_WIDGETS(AIF3TX4, "AIF3TX4"),
+
+MADERA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"),
+MADERA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"),
+MADERA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"),
+MADERA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"),
+MADERA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"),
+MADERA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"),
+MADERA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"),
+MADERA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"),
+
+MADERA_MUX_WIDGETS(SPD1TX1, "SPDIFTX1"),
+MADERA_MUX_WIDGETS(SPD1TX2, "SPDIFTX2"),
+
+MADERA_MUX_WIDGETS(ASRC1IN1L, "ASRC1IN1L"),
+MADERA_MUX_WIDGETS(ASRC1IN1R, "ASRC1IN1R"),
+MADERA_MUX_WIDGETS(ASRC1IN2L, "ASRC1IN2L"),
+MADERA_MUX_WIDGETS(ASRC1IN2R, "ASRC1IN2R"),
+
+MADERA_DSP_WIDGETS(DSP1, "DSP1"),
+
+MADERA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"),
+MADERA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"),
+
+MADERA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"),
+MADERA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"),
+
+MADERA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"),
+MADERA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
+
+MADERA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
+MADERA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
+
+MADERA_MUX_WIDGETS(DFC1, "DFC1"),
+MADERA_MUX_WIDGETS(DFC2, "DFC2"),
+MADERA_MUX_WIDGETS(DFC3, "DFC3"),
+MADERA_MUX_WIDGETS(DFC4, "DFC4"),
+MADERA_MUX_WIDGETS(DFC5, "DFC5"),
+MADERA_MUX_WIDGETS(DFC6, "DFC6"),
+MADERA_MUX_WIDGETS(DFC7, "DFC7"),
+MADERA_MUX_WIDGETS(DFC8, "DFC8"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2L"),
+SND_SOC_DAPM_OUTPUT("HPOUT2R"),
+SND_SOC_DAPM_OUTPUT("HPOUT3L"),
+SND_SOC_DAPM_OUTPUT("HPOUT3R"),
+SND_SOC_DAPM_OUTPUT("HPOUT4L"),
+SND_SOC_DAPM_OUTPUT("HPOUT4R"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+SND_SOC_DAPM_OUTPUT("SPDIF1"),
+
+SND_SOC_DAPM_OUTPUT("MICSUPP"),
+};
+
+#define MADERA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Noise Generator", "Noise Generator" }, \
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "Haptics", "HAPTICS" }, \
+	{ name, "AEC1", "AEC1 Loopback" }, \
+	{ name, "IN1L", "IN1L" }, \
+	{ name, "IN1R", "IN1R" }, \
+	{ name, "IN2L", "IN2L" }, \
+	{ name, "IN2R", "IN2R" }, \
+	{ name, "IN3L", "IN3L" }, \
+	{ name, "IN3R", "IN3R" }, \
+	{ name, "IN4L", "IN4L" }, \
+	{ name, "IN4R", "IN4R" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF1RX7", "AIF1RX7" }, \
+	{ name, "AIF1RX8", "AIF1RX8" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF2RX3", "AIF2RX3" }, \
+	{ name, "AIF2RX4", "AIF2RX4" }, \
+	{ name, "AIF2RX5", "AIF2RX5" }, \
+	{ name, "AIF2RX6", "AIF2RX6" }, \
+	{ name, "AIF2RX7", "AIF2RX7" }, \
+	{ name, "AIF2RX8", "AIF2RX8" }, \
+	{ name, "AIF3RX1", "AIF3RX1" }, \
+	{ name, "AIF3RX2", "AIF3RX2" }, \
+	{ name, "AIF3RX3", "AIF3RX3" }, \
+	{ name, "AIF3RX4", "AIF3RX4" }, \
+	{ name, "SLIMRX1", "SLIMRX1" }, \
+	{ name, "SLIMRX2", "SLIMRX2" }, \
+	{ name, "SLIMRX3", "SLIMRX3" }, \
+	{ name, "SLIMRX4", "SLIMRX4" }, \
+	{ name, "SLIMRX5", "SLIMRX5" }, \
+	{ name, "SLIMRX6", "SLIMRX6" }, \
+	{ name, "SLIMRX7", "SLIMRX7" }, \
+	{ name, "SLIMRX8", "SLIMRX8" }, \
+	{ name, "EQ1", "EQ1" }, \
+	{ name, "EQ2", "EQ2" }, \
+	{ name, "EQ3", "EQ3" }, \
+	{ name, "EQ4", "EQ4" }, \
+	{ name, "DRC1L", "DRC1L" }, \
+	{ name, "DRC1R", "DRC1R" }, \
+	{ name, "DRC2L", "DRC2L" }, \
+	{ name, "DRC2R", "DRC2R" }, \
+	{ name, "LHPF1", "LHPF1" }, \
+	{ name, "LHPF2", "LHPF2" }, \
+	{ name, "LHPF3", "LHPF3" }, \
+	{ name, "LHPF4", "LHPF4" }, \
+	{ name, "ASRC1IN1L", "ASRC1IN1L" }, \
+	{ name, "ASRC1IN1R", "ASRC1IN1R" }, \
+	{ name, "ASRC1IN2L", "ASRC1IN2L" }, \
+	{ name, "ASRC1IN2R", "ASRC1IN2R" }, \
+	{ name, "ISRC1DEC1", "ISRC1DEC1" }, \
+	{ name, "ISRC1DEC2", "ISRC1DEC2" }, \
+	{ name, "ISRC1INT1", "ISRC1INT1" }, \
+	{ name, "ISRC1INT2", "ISRC1INT2" }, \
+	{ name, "ISRC2DEC1", "ISRC2DEC1" }, \
+	{ name, "ISRC2DEC2", "ISRC2DEC2" }, \
+	{ name, "ISRC2INT1", "ISRC2INT1" }, \
+	{ name, "ISRC2INT2", "ISRC2INT2" }, \
+	{ name, "DSP1.1", "DSP1" }, \
+	{ name, "DSP1.2", "DSP1" }, \
+	{ name, "DSP1.3", "DSP1" }, \
+	{ name, "DSP1.4", "DSP1" }, \
+	{ name, "DSP1.5", "DSP1" }, \
+	{ name, "DSP1.6", "DSP1" }, \
+	{ name, "DFC1", "DFC1" }, \
+	{ name, "DFC2", "DFC2" }, \
+	{ name, "DFC3", "DFC3" }, \
+	{ name, "DFC4", "DFC4" }, \
+	{ name, "DFC5", "DFC5" }, \
+	{ name, "DFC6", "DFC6" }, \
+	{ name, "DFC7", "DFC7" }, \
+	{ name, "DFC8", "DFC8" }
+
+static const struct snd_soc_dapm_route cs47l92_dapm_routes[] = {
+	/* Internal clock domains */
+	{ "EQ1", NULL, "FXCLK" },
+	{ "EQ2", NULL, "FXCLK" },
+	{ "EQ3", NULL, "FXCLK" },
+	{ "EQ4", NULL, "FXCLK" },
+	{ "DRC1L", NULL, "FXCLK" },
+	{ "DRC1R", NULL, "FXCLK" },
+	{ "DRC2L", NULL, "FXCLK" },
+	{ "DRC2R", NULL, "FXCLK" },
+	{ "LHPF1", NULL, "FXCLK" },
+	{ "LHPF2", NULL, "FXCLK" },
+	{ "LHPF3", NULL, "FXCLK" },
+	{ "LHPF4", NULL, "FXCLK" },
+	{ "PWM1 Mixer", NULL, "PWMCLK" },
+	{ "PWM2 Mixer", NULL, "PWMCLK" },
+	{ "OUT1L", NULL, "OUTCLK" },
+	{ "OUT1R", NULL, "OUTCLK" },
+	{ "OUT2L", NULL, "OUTCLK" },
+	{ "OUT2R", NULL, "OUTCLK" },
+	{ "OUT3L", NULL, "OUTCLK" },
+	{ "OUT3R", NULL, "OUTCLK" },
+	{ "OUT5L", NULL, "OUTCLK" },
+	{ "OUT5R", NULL, "OUTCLK" },
+	{ "AIF1TX1", NULL, "AIF1TXCLK" },
+	{ "AIF1TX2", NULL, "AIF1TXCLK" },
+	{ "AIF1TX3", NULL, "AIF1TXCLK" },
+	{ "AIF1TX4", NULL, "AIF1TXCLK" },
+	{ "AIF1TX5", NULL, "AIF1TXCLK" },
+	{ "AIF1TX6", NULL, "AIF1TXCLK" },
+	{ "AIF1TX7", NULL, "AIF1TXCLK" },
+	{ "AIF1TX8", NULL, "AIF1TXCLK" },
+	{ "AIF2TX1", NULL, "AIF2TXCLK" },
+	{ "AIF2TX2", NULL, "AIF2TXCLK" },
+	{ "AIF2TX3", NULL, "AIF2TXCLK" },
+	{ "AIF2TX4", NULL, "AIF2TXCLK" },
+	{ "AIF2TX5", NULL, "AIF2TXCLK" },
+	{ "AIF2TX6", NULL, "AIF2TXCLK" },
+	{ "AIF2TX7", NULL, "AIF2TXCLK" },
+	{ "AIF2TX8", NULL, "AIF2TXCLK" },
+	{ "AIF3TX1", NULL, "AIF3TXCLK" },
+	{ "AIF3TX2", NULL, "AIF3TXCLK" },
+	{ "AIF3TX3", NULL, "AIF3TXCLK" },
+	{ "AIF3TX4", NULL, "AIF3TXCLK" },
+	{ "SLIMTX1", NULL, "SLIMBUSCLK" },
+	{ "SLIMTX2", NULL, "SLIMBUSCLK" },
+	{ "SLIMTX3", NULL, "SLIMBUSCLK" },
+	{ "SLIMTX4", NULL, "SLIMBUSCLK" },
+	{ "SLIMTX5", NULL, "SLIMBUSCLK" },
+	{ "SLIMTX6", NULL, "SLIMBUSCLK" },
+	{ "SLIMTX7", NULL, "SLIMBUSCLK" },
+	{ "SLIMTX8", NULL, "SLIMBUSCLK" },
+	{ "SPD1TX1", NULL, "SPDCLK" },
+	{ "SPD1TX2", NULL, "SPDCLK" },
+	{ "DSP1", NULL, "DSP1CLK" },
+	{ "ISRC1DEC1", NULL, "ISRC1CLK" },
+	{ "ISRC1DEC2", NULL, "ISRC1CLK" },
+	{ "ISRC1INT1", NULL, "ISRC1CLK" },
+	{ "ISRC1INT2", NULL, "ISRC1CLK" },
+	{ "ISRC2DEC1", NULL, "ISRC2CLK" },
+	{ "ISRC2DEC2", NULL, "ISRC2CLK" },
+	{ "ISRC2INT1", NULL, "ISRC2CLK" },
+	{ "ISRC2INT2", NULL, "ISRC2CLK" },
+	{ "ASRC1IN1L", NULL, "ASRC1CLK" },
+	{ "ASRC1IN1R", NULL, "ASRC1CLK" },
+	{ "ASRC1IN2L", NULL, "ASRC1CLK" },
+	{ "ASRC1IN2R", NULL, "ASRC1CLK" },
+	{ "DFC1", NULL, "DFCCLK" },
+	{ "DFC2", NULL, "DFCCLK" },
+	{ "DFC3", NULL, "DFCCLK" },
+	{ "DFC4", NULL, "DFCCLK" },
+	{ "DFC5", NULL, "DFCCLK" },
+	{ "DFC6", NULL, "DFCCLK" },
+	{ "DFC7", NULL, "DFCCLK" },
+	{ "DFC8", NULL, "DFCCLK" },
+
+	{ "OUT1L", NULL, "CPVDD1" },
+	{ "OUT1L", NULL, "CPVDD2" },
+	{ "OUT1R", NULL, "CPVDD1" },
+	{ "OUT1R", NULL, "CPVDD2" },
+	{ "OUT2L", NULL, "CPVDD1" },
+	{ "OUT2L", NULL, "CPVDD2" },
+	{ "OUT2R", NULL, "CPVDD1" },
+	{ "OUT2R", NULL, "CPVDD2" },
+	{ "OUT3L", NULL, "CPVDD1" },
+	{ "OUT3L", NULL, "CPVDD2" },
+	{ "OUT3R", NULL, "CPVDD1" },
+	{ "OUT3R", NULL, "CPVDD2" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT2L", NULL, "SYSCLK" },
+	{ "OUT2R", NULL, "SYSCLK" },
+	{ "OUT3L", NULL, "SYSCLK" },
+	{ "OUT3R", NULL, "SYSCLK" },
+	{ "OUT5L", NULL, "SYSCLK" },
+	{ "OUT5R", NULL, "SYSCLK" },
+
+	{ "SPD1", NULL, "SYSCLK" },
+	{ "SPD1", NULL, "SPD1TX1" },
+	{ "SPD1", NULL, "SPD1TX2" },
+
+	{ "IN1L", NULL, "SYSCLK" },
+	{ "IN1R", NULL, "SYSCLK" },
+	{ "IN2L", NULL, "SYSCLK" },
+	{ "IN2R", NULL, "SYSCLK" },
+	{ "IN3L", NULL, "SYSCLK" },
+	{ "IN3R", NULL, "SYSCLK" },
+	{ "IN4L", NULL, "SYSCLK" },
+	{ "IN4R", NULL, "SYSCLK" },
+
+	{ "ASRC1IN1L", NULL, "SYSCLK" },
+	{ "ASRC1IN1R", NULL, "SYSCLK" },
+	{ "ASRC1IN2L", NULL, "SYSCLK" },
+	{ "ASRC1IN2R", NULL, "SYSCLK" },
+
+	{ "ASRC1IN1L", NULL, "ASYNCCLK" },
+	{ "ASRC1IN1R", NULL, "ASYNCCLK" },
+	{ "ASRC1IN2L", NULL, "ASYNCCLK" },
+	{ "ASRC1IN2R", NULL, "ASYNCCLK" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+	{ "MICBIAS2", NULL, "MICVDD" },
+
+	{ "MICBIAS1A", NULL, "MICBIAS1" },
+	{ "MICBIAS1B", NULL, "MICBIAS1" },
+	{ "MICBIAS1C", NULL, "MICBIAS1" },
+	{ "MICBIAS1D", NULL, "MICBIAS1" },
+
+	{ "MICBIAS2A", NULL, "MICBIAS2" },
+	{ "MICBIAS2B", NULL, "MICBIAS2" },
+
+	{ "Noise Generator", NULL, "SYSCLK" },
+	{ "Tone Generator 1", NULL, "SYSCLK" },
+	{ "Tone Generator 2", NULL, "SYSCLK" },
+
+	{ "Noise Generator", NULL, "NOISE" },
+	{ "Tone Generator 1", NULL, "TONE" },
+	{ "Tone Generator 2", NULL, "TONE" },
+
+	{ "AIF1 Capture", NULL, "AIF1TX1" },
+	{ "AIF1 Capture", NULL, "AIF1TX2" },
+	{ "AIF1 Capture", NULL, "AIF1TX3" },
+	{ "AIF1 Capture", NULL, "AIF1TX4" },
+	{ "AIF1 Capture", NULL, "AIF1TX5" },
+	{ "AIF1 Capture", NULL, "AIF1TX6" },
+	{ "AIF1 Capture", NULL, "AIF1TX7" },
+	{ "AIF1 Capture", NULL, "AIF1TX8" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+	{ "AIF1RX7", NULL, "AIF1 Playback" },
+	{ "AIF1RX8", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+	{ "AIF2 Capture", NULL, "AIF2TX3" },
+	{ "AIF2 Capture", NULL, "AIF2TX4" },
+	{ "AIF2 Capture", NULL, "AIF2TX5" },
+	{ "AIF2 Capture", NULL, "AIF2TX6" },
+	{ "AIF2 Capture", NULL, "AIF2TX7" },
+	{ "AIF2 Capture", NULL, "AIF2TX8" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+	{ "AIF2RX3", NULL, "AIF2 Playback" },
+	{ "AIF2RX4", NULL, "AIF2 Playback" },
+	{ "AIF2RX5", NULL, "AIF2 Playback" },
+	{ "AIF2RX6", NULL, "AIF2 Playback" },
+	{ "AIF2RX7", NULL, "AIF2 Playback" },
+	{ "AIF2RX8", NULL, "AIF2 Playback" },
+
+	{ "AIF3 Capture", NULL, "AIF3TX1" },
+	{ "AIF3 Capture", NULL, "AIF3TX2" },
+	{ "AIF3 Capture", NULL, "AIF3TX3" },
+	{ "AIF3 Capture", NULL, "AIF3TX4" },
+
+	{ "AIF3RX1", NULL, "AIF3 Playback" },
+	{ "AIF3RX2", NULL, "AIF3 Playback" },
+	{ "AIF3RX3", NULL, "AIF3 Playback" },
+	{ "AIF3RX4", NULL, "AIF3 Playback" },
+
+	{ "Slim1 Capture", NULL, "SLIMTX1" },
+	{ "Slim1 Capture", NULL, "SLIMTX2" },
+	{ "Slim1 Capture", NULL, "SLIMTX3" },
+	{ "Slim1 Capture", NULL, "SLIMTX4" },
+
+	{ "SLIMRX1", NULL, "Slim1 Playback" },
+	{ "SLIMRX2", NULL, "Slim1 Playback" },
+	{ "SLIMRX3", NULL, "Slim1 Playback" },
+	{ "SLIMRX4", NULL, "Slim1 Playback" },
+
+	{ "Slim2 Capture", NULL, "SLIMTX5" },
+	{ "Slim2 Capture", NULL, "SLIMTX6" },
+
+	{ "SLIMRX5", NULL, "Slim2 Playback" },
+	{ "SLIMRX6", NULL, "Slim2 Playback" },
+
+	{ "Slim3 Capture", NULL, "SLIMTX7" },
+	{ "Slim3 Capture", NULL, "SLIMTX8" },
+
+	{ "SLIMRX7", NULL, "Slim3 Playback" },
+	{ "SLIMRX8", NULL, "Slim3 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "AIF3 Playback", NULL, "SYSCLK" },
+	{ "Slim1 Playback", NULL, "SYSCLK" },
+	{ "Slim2 Playback", NULL, "SYSCLK" },
+	{ "Slim3 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "AIF3 Capture", NULL, "SYSCLK" },
+	{ "Slim1 Capture", NULL, "SYSCLK" },
+	{ "Slim2 Capture", NULL, "SYSCLK" },
+	{ "Slim3 Capture", NULL, "SYSCLK" },
+
+	{ "Audio Trace DSP", NULL, "DSP1" },
+
+	{ "IN1L Analog Mux", "A", "IN1ALN" },
+	{ "IN1L Analog Mux", "A", "IN1ALP" },
+	{ "IN1L Analog Mux", "B", "IN1BLN" },
+	{ "IN1L Analog Mux", "B", "IN1BLP" },
+	{ "IN1R Analog Mux", "A", "IN1ARN" },
+	{ "IN1R Analog Mux", "A", "IN1ARP" },
+	{ "IN1R Analog Mux", "B", "IN1BR" },
+	{ "IN1R Analog Mux", "B", "IN1ALN" },
+
+	{ "IN1L Mode", "Analog", "IN1L Analog Mux" },
+	{ "IN1R Mode", "Analog", "IN1R Analog Mux" },
+
+	{ "IN1L Mode", "Digital", "IN1ALN" },
+	{ "IN1L Mode", "Digital", "IN1ALP" },
+	{ "IN1R Mode", "Digital", "IN1ALN" },
+	{ "IN1R Mode", "Digital", "IN1ALP" },
+
+	{ "IN1L", NULL, "IN1L Mode" },
+	{ "IN1R", NULL, "IN1R Mode" },
+
+	{ "IN2L Analog Mux", "A", "IN2ALN" },
+	{ "IN2L Analog Mux", "A", "IN2ALP" },
+	{ "IN2L Analog Mux", "B", "IN2ALN" },
+	{ "IN2L Analog Mux", "B", "IN2BL" },
+	{ "IN2R Analog Mux", "A", "IN2ARN" },
+	{ "IN2R Analog Mux", "A", "IN2ARP" },
+	{ "IN2R Analog Mux", "B", "IN2ARN" },
+	{ "IN2R Analog Mux", "B", "IN2BR" },
+
+	{ "IN2L Mode", "Analog", "IN2L Analog Mux" },
+	{ "IN2R Mode", "Analog", "IN2R Analog Mux" },
+
+	{ "IN2L Mode", "Digital", "IN2ALN" },
+	{ "IN2L Mode", "Digital", "IN2ALP" },
+	{ "IN2R Mode", "Digital", "IN2ALN" },
+	{ "IN2R Mode", "Digital", "IN2ALP" },
+
+	{ "IN2L", NULL, "IN2L Mode" },
+	{ "IN2R", NULL, "IN2R Mode" },
+
+	{ "IN3L", NULL, "IN1ARN" },
+	{ "IN3L", NULL, "IN1ARP" },
+	{ "IN3R", NULL, "IN1ARN" },
+	{ "IN3R", NULL, "IN1ARP" },
+
+	{ "IN4L", NULL, "IN2ARN" },
+	{ "IN4L", NULL, "IN2ARP" },
+	{ "IN4R", NULL, "IN2ARN" },
+	{ "IN4R", NULL, "IN2ARP" },
+
+	MADERA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+	MADERA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+	MADERA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
+	MADERA_MIXER_ROUTES("OUT2R", "HPOUT2R"),
+	MADERA_MIXER_ROUTES("OUT3L", "HPOUT3L"),
+	MADERA_MIXER_ROUTES("OUT3R", "HPOUT3R"),
+
+	MADERA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+	MADERA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+
+	MADERA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+	MADERA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+	MADERA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+	MADERA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+	MADERA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+	MADERA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+	MADERA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+	MADERA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+	MADERA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+	MADERA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+	MADERA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+	MADERA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+	MADERA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"),
+	MADERA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"),
+	MADERA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"),
+	MADERA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"),
+	MADERA_MIXER_ROUTES("AIF2TX7", "AIF2TX7"),
+	MADERA_MIXER_ROUTES("AIF2TX8", "AIF2TX8"),
+
+	MADERA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+	MADERA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+	MADERA_MIXER_ROUTES("AIF3TX3", "AIF3TX3"),
+	MADERA_MIXER_ROUTES("AIF3TX4", "AIF3TX4"),
+
+	MADERA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"),
+	MADERA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"),
+	MADERA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"),
+	MADERA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"),
+	MADERA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"),
+	MADERA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"),
+	MADERA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"),
+	MADERA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"),
+
+	MADERA_MUX_ROUTES("SPD1TX1", "SPDIFTX1"),
+	MADERA_MUX_ROUTES("SPD1TX2", "SPDIFTX2"),
+
+	MADERA_MIXER_ROUTES("EQ1", "EQ1"),
+	MADERA_MIXER_ROUTES("EQ2", "EQ2"),
+	MADERA_MIXER_ROUTES("EQ3", "EQ3"),
+	MADERA_MIXER_ROUTES("EQ4", "EQ4"),
+
+	MADERA_MIXER_ROUTES("DRC1L", "DRC1L"),
+	MADERA_MIXER_ROUTES("DRC1R", "DRC1R"),
+	MADERA_MIXER_ROUTES("DRC2L", "DRC2L"),
+	MADERA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+	MADERA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	MADERA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	MADERA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	MADERA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	MADERA_MUX_ROUTES("ASRC1IN1L", "ASRC1IN1L"),
+	MADERA_MUX_ROUTES("ASRC1IN1R", "ASRC1IN1R"),
+	MADERA_MUX_ROUTES("ASRC1IN2L", "ASRC1IN2L"),
+	MADERA_MUX_ROUTES("ASRC1IN2R", "ASRC1IN2R"),
+
+	MADERA_DSP_ROUTES("DSP1"),
+
+	MADERA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+	MADERA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"),
+
+	MADERA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+	MADERA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
+
+	MADERA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+	MADERA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+
+	MADERA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+	MADERA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
+
+	{ "AEC1 Loopback", "HPOUT1L", "OUT1L" },
+	{ "AEC1 Loopback", "HPOUT1R", "OUT1R" },
+	{ "HPOUT1L", NULL, "OUT1L" },
+	{ "HPOUT1R", NULL, "OUT1R" },
+
+	{ "AEC1 Loopback", "HPOUT2L", "OUT2L" },
+	{ "AEC1 Loopback", "HPOUT2R", "OUT2R" },
+	{ "HPOUT2L", NULL, "OUT2L" },
+	{ "HPOUT2R", NULL, "OUT2R" },
+
+	{ "AEC1 Loopback", "HPOUT3L", "OUT3L" },
+	{ "AEC1 Loopback", "HPOUT3R", "OUT3R" },
+	{ "OUT3 Demux", NULL, "OUT3L" },
+	{ "OUT3 Demux", NULL, "OUT3R" },
+
+	{ "HPOUT3L", "HPOUT3", "OUT3 Demux" },
+	{ "HPOUT3R", "HPOUT3", "OUT3 Demux" },
+	{ "HPOUT4L", "HPOUT4", "OUT3 Demux" },
+	{ "HPOUT4R", "HPOUT4", "OUT3 Demux" },
+
+	{ "AEC1 Loopback", "SPKDAT1L", "OUT5L" },
+	{ "AEC1 Loopback", "SPKDAT1R", "OUT5R" },
+	{ "SPKDAT1L", NULL, "OUT5L" },
+	{ "SPKDAT1R", NULL, "OUT5R" },
+
+	{ "SPDIF1", NULL, "SPD1" },
+
+	{ "MICSUPP", NULL, "SYSCLK" },
+
+	{ "DRC1 Signal Activity", NULL, "DRC1 Activity Output" },
+	{ "DRC2 Signal Activity", NULL, "DRC2 Activity Output" },
+	{ "DRC1 Activity Output", "Switch", "DRC1L" },
+	{ "DRC1 Activity Output", "Switch", "DRC1R" },
+	{ "DRC2 Activity Output", "Switch", "DRC2L" },
+	{ "DRC2 Activity Output", "Switch", "DRC2R" },
+
+	MADERA_MUX_ROUTES("DFC1", "DFC1"),
+	MADERA_MUX_ROUTES("DFC2", "DFC2"),
+	MADERA_MUX_ROUTES("DFC3", "DFC3"),
+	MADERA_MUX_ROUTES("DFC4", "DFC4"),
+	MADERA_MUX_ROUTES("DFC5", "DFC5"),
+	MADERA_MUX_ROUTES("DFC6", "DFC6"),
+	MADERA_MUX_ROUTES("DFC7", "DFC7"),
+	MADERA_MUX_ROUTES("DFC8", "DFC8"),
+};
+
+static int cs47l92_set_fll(struct snd_soc_component *component, int fll_id,
+			   int source, unsigned int fref, unsigned int fout)
+{
+	struct cs47l92 *cs47l92 = snd_soc_component_get_drvdata(component);
+
+	switch (fll_id) {
+	case MADERA_FLL1_REFCLK:
+		return madera_fllhj_set_refclk(&cs47l92->fll[0], source, fref,
+					       fout);
+	case MADERA_FLL2_REFCLK:
+		return madera_fllhj_set_refclk(&cs47l92->fll[1], source, fref,
+					       fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+static struct snd_soc_dai_driver cs47l92_dai[] = {
+	{
+		.name = "cs47l92-aif1",
+		.id = 1,
+		.base = MADERA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF1 Capture",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "cs47l92-aif2",
+		.id = 2,
+		.base = MADERA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF2 Capture",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "cs47l92-aif3",
+		.id = 3,
+		.base = MADERA_AIF3_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF3 Capture",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "cs47l92-slim1",
+		.id = 5,
+		.playback = {
+			.stream_name = "Slim1 Playback",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Slim1 Capture",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_simple_dai_ops,
+	},
+	{
+		.name = "cs47l92-slim2",
+		.id = 6,
+		.playback = {
+			.stream_name = "Slim2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Slim2 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_simple_dai_ops,
+	},
+	{
+		.name = "cs47l92-slim3",
+		.id = 7,
+		.playback = {
+			.stream_name = "Slim3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Slim3 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		 },
+		.ops = &madera_simple_dai_ops,
+	},
+	{
+		.name = "cs47l92-cpu-trace",
+		.capture = {
+			.stream_name = "Audio Trace CPU",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+		.compress_new = snd_soc_new_compress,
+	},
+	{
+		.name = "cs47l92-dsp-trace",
+		.capture = {
+			.stream_name = "Audio Trace DSP",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MADERA_RATES,
+			.formats = MADERA_FORMATS,
+		},
+	},
+};
+
+static int cs47l92_open(struct snd_compr_stream *stream)
+{
+	struct snd_soc_pcm_runtime *rtd = stream->private_data;
+	struct snd_soc_component *component =
+		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+	struct cs47l92 *cs47l92 = snd_soc_component_get_drvdata(component);
+	struct madera_priv *priv = &cs47l92->core;
+	struct madera *madera = priv->madera;
+	int n_adsp;
+
+	if (strcmp(rtd->codec_dai->name, "cs47l92-dsp-trace") == 0) {
+		n_adsp = 0;
+	} else {
+		dev_err(madera->dev,
+			"No suitable compressed stream for DAI '%s'\n",
+			rtd->codec_dai->name);
+		return -EINVAL;
+	}
+
+	return wm_adsp_compr_open(&priv->adsp[n_adsp], stream);
+}
+
+static irqreturn_t cs47l92_adsp2_irq(int irq, void *data)
+{
+	struct cs47l92 *cs47l92 = data;
+	struct madera_priv *priv = &cs47l92->core;
+	struct madera *madera = priv->madera;
+	int ret;
+
+	ret = wm_adsp_compr_handle_irq(&priv->adsp[0]);
+	if (ret == -ENODEV) {
+		dev_err(madera->dev, "Spurious compressed data IRQ\n");
+		return IRQ_NONE;
+	}
+
+	return IRQ_HANDLED;
+}
+
+static int cs47l92_component_probe(struct snd_soc_component *component)
+{
+	struct cs47l92 *cs47l92 = snd_soc_component_get_drvdata(component);
+	struct madera *madera = cs47l92->core.madera;
+	int ret;
+
+	snd_soc_component_init_regmap(component, madera->regmap);
+
+	mutex_lock(&madera->dapm_ptr_lock);
+	madera->dapm = snd_soc_component_get_dapm(component);
+	mutex_unlock(&madera->dapm_ptr_lock);
+
+	ret = madera_init_inputs(component);
+	if (ret)
+		return ret;
+
+	ret = madera_init_outputs(component, CS47L92_MONO_OUTPUTS);
+	if (ret)
+		return ret;
+
+	snd_soc_component_disable_pin(component, "HAPTICS");
+
+	ret = snd_soc_add_component_controls(component,
+					     madera_adsp_rate_controls,
+					     CS47L92_NUM_ADSP);
+	if (ret)
+		return ret;
+
+	return wm_adsp2_component_probe(&cs47l92->core.adsp[0], component);
+}
+
+static void cs47l92_component_remove(struct snd_soc_component *component)
+{
+	struct cs47l92 *cs47l92 = snd_soc_component_get_drvdata(component);
+	struct madera *madera = cs47l92->core.madera;
+
+	mutex_lock(&madera->dapm_ptr_lock);
+	madera->dapm = NULL;
+	mutex_unlock(&madera->dapm_ptr_lock);
+
+	wm_adsp2_component_remove(&cs47l92->core.adsp[0], component);
+}
+
+#define CS47L92_DIG_VU 0x0200
+
+static unsigned int cs47l92_digital_vu[] = {
+	MADERA_DAC_DIGITAL_VOLUME_1L,
+	MADERA_DAC_DIGITAL_VOLUME_1R,
+	MADERA_DAC_DIGITAL_VOLUME_2L,
+	MADERA_DAC_DIGITAL_VOLUME_2R,
+	MADERA_DAC_DIGITAL_VOLUME_3L,
+	MADERA_DAC_DIGITAL_VOLUME_3R,
+	MADERA_DAC_DIGITAL_VOLUME_5L,
+	MADERA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static const struct snd_compr_ops cs47l92_compr_ops = {
+	.open = &cs47l92_open,
+	.free = &wm_adsp_compr_free,
+	.set_params = &wm_adsp_compr_set_params,
+	.get_caps = &wm_adsp_compr_get_caps,
+	.trigger = &wm_adsp_compr_trigger,
+	.pointer = &wm_adsp_compr_pointer,
+	.copy = &wm_adsp_compr_copy,
+};
+
+static const struct snd_soc_component_driver soc_component_dev_cs47l92 = {
+	.probe			= &cs47l92_component_probe,
+	.remove			= &cs47l92_component_remove,
+	.set_sysclk		= &madera_set_sysclk,
+	.set_pll		= &cs47l92_set_fll,
+	.name			= DRV_NAME,
+	.compr_ops		= &cs47l92_compr_ops,
+	.controls		= cs47l92_snd_controls,
+	.num_controls		= ARRAY_SIZE(cs47l92_snd_controls),
+	.dapm_widgets		= cs47l92_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(cs47l92_dapm_widgets),
+	.dapm_routes		= cs47l92_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(cs47l92_dapm_routes),
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static int cs47l92_probe(struct platform_device *pdev)
+{
+	struct madera *madera = dev_get_drvdata(pdev->dev.parent);
+	struct cs47l92 *cs47l92;
+	int i, ret;
+
+	BUILD_BUG_ON(ARRAY_SIZE(cs47l92_dai) > MADERA_MAX_DAI);
+
+	/* quick exit if Madera irqchip driver hasn't completed probe */
+	if (!madera->irq_dev) {
+		dev_dbg(&pdev->dev, "irqchip driver not ready\n");
+		return -EPROBE_DEFER;
+	}
+
+	cs47l92 = devm_kzalloc(&pdev->dev, sizeof(struct cs47l92), GFP_KERNEL);
+	if (!cs47l92)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, cs47l92);
+
+	cs47l92->core.madera = madera;
+	cs47l92->core.dev = &pdev->dev;
+	cs47l92->core.num_inputs = 8;
+
+	ret = madera_core_init(&cs47l92->core);
+	if (ret)
+		return ret;
+
+	ret = madera_request_irq(madera, MADERA_IRQ_DSP_IRQ1,
+				 "ADSP2 Compressed IRQ", cs47l92_adsp2_irq,
+				 cs47l92);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to request DSP IRQ: %d\n", ret);
+		goto error_core;
+	}
+
+	ret = madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 1);
+	if (ret)
+		dev_warn(&pdev->dev, "Failed to set DSP IRQ wake: %d\n", ret);
+
+	cs47l92->core.adsp[0].part = "cs47l92";
+	cs47l92->core.adsp[0].num = 1;
+	cs47l92->core.adsp[0].type = WMFW_ADSP2;
+	cs47l92->core.adsp[0].rev = 2;
+	cs47l92->core.adsp[0].dev = madera->dev;
+	cs47l92->core.adsp[0].regmap = madera->regmap_32bit;
+
+	cs47l92->core.adsp[0].base = MADERA_DSP1_CONFIG_1;
+	cs47l92->core.adsp[0].mem = cs47l92_dsp1_regions;
+	cs47l92->core.adsp[0].num_mems = ARRAY_SIZE(cs47l92_dsp1_regions);
+
+	cs47l92->core.adsp[0].lock_regions = WM_ADSP2_REGION_1_9;
+
+	ret = wm_adsp2_init(&cs47l92->core.adsp[0]);
+	if (ret != 0)
+		goto error_dsp_irq;
+
+	ret = madera_init_bus_error_irq(&cs47l92->core, 0, wm_adsp2_bus_error);
+	if (ret != 0) {
+		wm_adsp2_remove(&cs47l92->core.adsp[0]);
+		goto error_adsp;
+	}
+
+	madera_init_fll(madera, 1, MADERA_FLL1_CONTROL_1 - 1,
+			&cs47l92->fll[0]);
+	madera_init_fll(madera, 2, MADERA_FLL2_CONTROL_1 - 1,
+			&cs47l92->fll[1]);
+
+	for (i = 0; i < ARRAY_SIZE(cs47l92_dai); i++)
+		madera_init_dai(&cs47l92->core, i);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(cs47l92_digital_vu); i++)
+		regmap_update_bits(madera->regmap, cs47l92_digital_vu[i],
+				   CS47L92_DIG_VU, CS47L92_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &soc_component_dev_cs47l92,
+					      cs47l92_dai,
+					      ARRAY_SIZE(cs47l92_dai));
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register component: %d\n", ret);
+		goto error_pm_runtime;
+	}
+
+	return ret;
+
+error_pm_runtime:
+	pm_runtime_disable(&pdev->dev);
+	madera_free_bus_error_irq(&cs47l92->core, 0);
+error_adsp:
+	wm_adsp2_remove(&cs47l92->core.adsp[0]);
+error_dsp_irq:
+	madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 0);
+	madera_free_irq(madera, MADERA_IRQ_DSP_IRQ1, cs47l92);
+error_core:
+	madera_core_free(&cs47l92->core);
+
+	return ret;
+}
+
+static int cs47l92_remove(struct platform_device *pdev)
+{
+	struct cs47l92 *cs47l92 = platform_get_drvdata(pdev);
+
+	pm_runtime_disable(&pdev->dev);
+
+	madera_free_bus_error_irq(&cs47l92->core, 0);
+	wm_adsp2_remove(&cs47l92->core.adsp[0]);
+
+	madera_set_irq_wake(cs47l92->core.madera, MADERA_IRQ_DSP_IRQ1, 0);
+	madera_free_irq(cs47l92->core.madera, MADERA_IRQ_DSP_IRQ1, cs47l92);
+
+	madera_core_free(&cs47l92->core);
+
+	return 0;
+}
+
+static struct platform_driver cs47l92_codec_driver = {
+	.driver = {
+		.name = "cs47l92-codec",
+	},
+	.probe = &cs47l92_probe,
+	.remove = &cs47l92_remove,
+};
+
+module_platform_driver(cs47l92_codec_driver);
+
+MODULE_SOFTDEP("pre: madera irq-madera arizona-micsupp");
+MODULE_DESCRIPTION("ASoC CS47L92 driver");
+MODULE_AUTHOR("Stuart Henderson <stuarth@opensource.cirrus.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cs47l92-codec");
diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c
index 1c00e2a115bc..52639811cc52 100644
--- a/sound/soc/codecs/madera.c
+++ b/sound/soc/codecs/madera.c
@@ -87,6 +87,16 @@
 #define MADERA_FLLAO_MIN_N			4
 #define MADERA_FLLAO_MAX_N			1023
 #define MADERA_FLLAO_MAX_FBDIV			254
+#define MADERA_FLLHJ_INT_MAX_N			1023
+#define MADERA_FLLHJ_INT_MIN_N			1
+#define MADERA_FLLHJ_FRAC_MAX_N			255
+#define MADERA_FLLHJ_FRAC_MIN_N			4
+#define MADERA_FLLHJ_LOW_THRESH			192000
+#define MADERA_FLLHJ_MID_THRESH			1152000
+#define MADERA_FLLHJ_MAX_THRESH			13000000
+#define MADERA_FLLHJ_LOW_GAINS			0x23f0
+#define MADERA_FLLHJ_MID_GAINS			0x22f2
+#define MADERA_FLLHJ_HIGH_GAINS			0x21f0
 
 #define MADERA_FLL_SYNCHRONISER_OFFS		0x10
 #define CS47L35_FLL_SYNCHRONISER_OFFS		0xE
@@ -96,6 +106,7 @@
 #define MADERA_FLL_CONTROL_4_OFFS		0x4
 #define MADERA_FLL_CONTROL_5_OFFS		0x5
 #define MADERA_FLL_CONTROL_6_OFFS		0x6
+#define MADERA_FLL_GAIN_OFFS			0x8
 #define MADERA_FLL_CONTROL_7_OFFS		0x9
 #define MADERA_FLL_EFS_2_OFFS			0xA
 #define MADERA_FLL_SYNCHRONISER_1_OFFS		0x1
@@ -107,6 +118,9 @@
 #define MADERA_FLL_SYNCHRONISER_7_OFFS		0x7
 #define MADERA_FLL_SPREAD_SPECTRUM_OFFS		0x9
 #define MADERA_FLL_GPIO_CLOCK_OFFS		0xA
+#define MADERA_FLL_CONTROL_10_OFFS		0xA
+#define MADERA_FLL_CONTROL_11_OFFS		0xB
+#define MADERA_FLL1_DIGITAL_TEST_1_OFFS		0xD
 
 #define MADERA_FLLAO_CONTROL_1_OFFS		0x1
 #define MADERA_FLLAO_CONTROL_2_OFFS		0x2
@@ -1871,6 +1885,18 @@ const struct soc_enum madera_asrc1_rate[] = {
 };
 EXPORT_SYMBOL_GPL(madera_asrc1_rate);
 
+const struct soc_enum madera_asrc1_bidir_rate[] = {
+	SOC_VALUE_ENUM_SINGLE(MADERA_ASRC1_RATE1,
+			      MADERA_ASRC1_RATE1_SHIFT, 0xf,
+			      MADERA_RATE_ENUM_SIZE,
+			      madera_rate_text, madera_rate_val),
+	SOC_VALUE_ENUM_SINGLE(MADERA_ASRC1_RATE2,
+			      MADERA_ASRC1_RATE2_SHIFT, 0xf,
+			      MADERA_RATE_ENUM_SIZE,
+			      madera_rate_text, madera_rate_val),
+};
+EXPORT_SYMBOL_GPL(madera_asrc1_bidir_rate);
+
 const struct soc_enum madera_asrc2_rate[] = {
 	SOC_VALUE_ENUM_SINGLE(MADERA_ASRC2_RATE1,
 			      MADERA_ASRC2_RATE1_SHIFT, 0xf,
@@ -2250,6 +2276,9 @@ int madera_out_ev(struct snd_soc_dapm_widget *w,
 	switch (madera->type) {
 	case CS47L90:
 	case CS47L91:
+	case CS42L92:
+	case CS47L92:
+	case CS47L93:
 		out_up_delay = 6;
 		break;
 	default:
@@ -2365,9 +2394,17 @@ int madera_hp_ev(struct snd_soc_dapm_widget *w,
 	madera->hp_ena &= ~mask;
 	madera->hp_ena |= val;
 
-	/* if OUT1 is routed to EPOUT, ignore HP clamp and impedance */
-	regmap_read(madera->regmap, MADERA_OUTPUT_ENABLES_1, &ep_sel);
-	ep_sel &= MADERA_EP_SEL_MASK;
+	switch (madera->type) {
+	case CS42L92:
+	case CS47L92:
+	case CS47L93:
+		break;
+	default:
+		/* if OUT1 is routed to EPOUT, ignore HP clamp and impedance */
+		regmap_read(madera->regmap, MADERA_OUTPUT_ENABLES_1, &ep_sel);
+		ep_sel &= MADERA_EP_SEL_MASK;
+		break;
+	}
 
 	/* Force off if HPDET has disabled the clamp for this output */
 	if (!ep_sel &&
@@ -2543,6 +2580,58 @@ static int madera_get_dspclk_setting(struct madera *madera,
 	}
 }
 
+static int madera_set_outclk(struct snd_soc_component *component,
+			     unsigned int source, unsigned int freq)
+{
+	int div, div_inc, rate;
+
+	switch (source) {
+	case MADERA_OUTCLK_SYSCLK:
+		dev_dbg(component->dev, "Configured OUTCLK to SYSCLK\n");
+		snd_soc_component_update_bits(component, MADERA_OUTPUT_RATE_1,
+					      MADERA_OUT_CLK_SRC_MASK, source);
+		return 0;
+	case MADERA_OUTCLK_ASYNCCLK:
+		dev_dbg(component->dev, "Configured OUTCLK to ASYNCCLK\n");
+		snd_soc_component_update_bits(component, MADERA_OUTPUT_RATE_1,
+					      MADERA_OUT_CLK_SRC_MASK, source);
+		return 0;
+	case MADERA_OUTCLK_MCLK1:
+	case MADERA_OUTCLK_MCLK2:
+	case MADERA_OUTCLK_MCLK3:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (freq % 4000)
+		rate = 5644800;
+	else
+		rate = 6144000;
+
+	div = 1;
+	div_inc = 0;
+	while (div <= 8) {
+		if (freq / div == rate && !(freq % div)) {
+			dev_dbg(component->dev, "Configured %dHz OUTCLK\n", rate);
+			snd_soc_component_update_bits(component,
+				MADERA_OUTPUT_RATE_1,
+				MADERA_OUT_EXT_CLK_DIV_MASK |
+				MADERA_OUT_CLK_SRC_MASK,
+				(div_inc << MADERA_OUT_EXT_CLK_DIV_SHIFT) |
+				source);
+			return 0;
+		}
+		div_inc++;
+		div *= 2;
+	}
+
+	dev_err(component->dev,
+		"Unable to generate %dHz OUTCLK from %dHz MCLK\n",
+		rate, freq);
+	return -EINVAL;
+}
+
 int madera_set_sysclk(struct snd_soc_component *component, int clk_id,
 		      int source, unsigned int freq, int dir)
 {
@@ -2579,6 +2668,8 @@ int madera_set_sysclk(struct snd_soc_component *component, int clk_id,
 	case MADERA_CLK_OPCLK:
 	case MADERA_CLK_ASYNC_OPCLK:
 		return madera_set_opclk(component, clk_id, freq);
+	case MADERA_CLK_OUTCLK:
+		return madera_set_outclk(component, source, freq);
 	default:
 		return -EINVAL;
 	}
@@ -2792,6 +2883,10 @@ static const unsigned int madera_sr_vals[] = {
 #define MADERA_192K_44K1_RATE_MASK	0x003E00
 #define MADERA_192K_RATE_MASK		(MADERA_192K_48K_RATE_MASK | \
 					 MADERA_192K_44K1_RATE_MASK)
+#define MADERA_384K_48K_RATE_MASK	0x0F007E
+#define MADERA_384K_44K1_RATE_MASK	0x007E00
+#define MADERA_384K_RATE_MASK		(MADERA_384K_48K_RATE_MASK | \
+					 MADERA_384K_44K1_RATE_MASK)
 
 static const struct snd_pcm_hw_constraint_list madera_constraint = {
 	.count	= ARRAY_SIZE(madera_sr_vals),
@@ -2804,6 +2899,7 @@ static int madera_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct madera_priv *priv = snd_soc_component_get_drvdata(component);
 	struct madera_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+	struct madera *madera = priv->madera;
 	unsigned int base_rate;
 
 	if (!substream->runtime)
@@ -2823,12 +2919,26 @@ static int madera_startup(struct snd_pcm_substream *substream,
 		return 0;
 	}
 
-	if (base_rate == 0)
-		dai_priv->constraint.mask = MADERA_192K_RATE_MASK;
-	else if (base_rate % 4000)
-		dai_priv->constraint.mask = MADERA_192K_44K1_RATE_MASK;
-	else
-		dai_priv->constraint.mask = MADERA_192K_48K_RATE_MASK;
+	switch (madera->type) {
+	case CS42L92:
+	case CS47L92:
+	case CS47L93:
+		if (base_rate == 0)
+			dai_priv->constraint.mask = MADERA_384K_RATE_MASK;
+		else if (base_rate % 4000)
+			dai_priv->constraint.mask = MADERA_384K_44K1_RATE_MASK;
+		else
+			dai_priv->constraint.mask = MADERA_384K_48K_RATE_MASK;
+		break;
+	default:
+		if (base_rate == 0)
+			dai_priv->constraint.mask = MADERA_192K_RATE_MASK;
+		else if (base_rate % 4000)
+			dai_priv->constraint.mask = MADERA_192K_44K1_RATE_MASK;
+		else
+			dai_priv->constraint.mask = MADERA_192K_48K_RATE_MASK;
+		break;
+	}
 
 	return snd_pcm_hw_constraint_list(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_RATE,
@@ -4149,6 +4259,308 @@ int madera_set_fll_ao_refclk(struct madera_fll *fll, int source,
 }
 EXPORT_SYMBOL_GPL(madera_set_fll_ao_refclk);
 
+static int madera_fllhj_disable(struct madera_fll *fll)
+{
+	struct madera *madera = fll->madera;
+	bool change;
+
+	madera_fll_dbg(fll, "Disabling FLL\n");
+
+	/* Disable lockdet, but don't set ctrl_upd update but.  This allows the
+	 * lock status bit to clear as normal, but should the FLL be enabled
+	 * again due to a control clock being required, the lock won't re-assert
+	 * as the FLL config registers are automatically applied when the FLL
+	 * enables.
+	 */
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_11_OFFS,
+			   MADERA_FLL1_LOCKDET_MASK, 0);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_1_OFFS,
+			   MADERA_FLL1_HOLD_MASK, MADERA_FLL1_HOLD_MASK);
+	regmap_update_bits_check(madera->regmap,
+				 fll->base + MADERA_FLL_CONTROL_1_OFFS,
+				 MADERA_FLL1_ENA_MASK, 0, &change);
+
+	madera_wait_for_fll(fll, false);
+
+	/* ctrl_up gates the writes to all the fll's registers, setting it to 0
+	 * here ensures that after a runtime suspend/resume cycle when one
+	 * enables the fll then ctrl_up is the last bit that is configured
+	 * by the fll enable code rather than the cache sync operation which
+	 * would have updated it much earlier before writing out all fll
+	 * registers
+	 */
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_2_OFFS,
+			   MADERA_FLL1_CTRL_UPD_MASK, 0);
+
+	if (change)
+		pm_runtime_put_autosuspend(madera->dev);
+
+	return 0;
+}
+
+static int madera_fllhj_apply(struct madera_fll *fll, int fin)
+{
+	struct madera *madera = fll->madera;
+	int refdiv, fref, fout, lockdet_thr, fbdiv, hp, fast_clk, fllgcd;
+	bool frac = false;
+	unsigned int fll_n, min_n, max_n, ratio, theta, lambda;
+	unsigned int gains, val, num;
+
+	madera_fll_dbg(fll, "fin=%d, fout=%d\n", fin, fll->fout);
+
+	for (refdiv = 0; refdiv < 4; refdiv++)
+		if ((fin / (1 << refdiv)) <= MADERA_FLLHJ_MAX_THRESH)
+			break;
+
+	fref = fin / (1 << refdiv);
+
+	/* Use simple heuristic approach to find a configuration that
+	 * should work for most input clocks.
+	 */
+	fast_clk = 0;
+	fout = fll->fout;
+	frac = fout % fref;
+
+	if (fref < MADERA_FLLHJ_LOW_THRESH) {
+		lockdet_thr = 2;
+		gains = MADERA_FLLHJ_LOW_GAINS;
+		if (frac)
+			fbdiv = 256;
+		else
+			fbdiv = 4;
+	} else if (fref < MADERA_FLLHJ_MID_THRESH) {
+		lockdet_thr = 8;
+		gains = MADERA_FLLHJ_MID_GAINS;
+		fbdiv = 1;
+	} else {
+		lockdet_thr = 8;
+		gains = MADERA_FLLHJ_HIGH_GAINS;
+		fbdiv = 1;
+		/* For high speed input clocks, enable 300MHz fast oscillator
+		 * when we're in fractional divider mode.
+		 */
+		if (frac) {
+			fast_clk = 0x3;
+			fout = fll->fout * 6;
+		}
+	}
+	/* Use high performance mode for fractional configurations. */
+	if (frac) {
+		hp = 0x3;
+		min_n = MADERA_FLLHJ_FRAC_MIN_N;
+		max_n = MADERA_FLLHJ_FRAC_MAX_N;
+	} else {
+		hp = 0x0;
+		min_n = MADERA_FLLHJ_INT_MIN_N;
+		max_n = MADERA_FLLHJ_INT_MAX_N;
+	}
+
+	ratio = fout / fref;
+
+	madera_fll_dbg(fll, "refdiv=%d, fref=%d, frac:%d\n",
+		       refdiv, fref, frac);
+
+	while (ratio / fbdiv < min_n) {
+		fbdiv /= 2;
+		if (fbdiv < 1) {
+			madera_fll_err(fll, "FBDIV (%d) must be >= 1\n", fbdiv);
+			return -EINVAL;
+		}
+	}
+	while (frac && (ratio / fbdiv > max_n)) {
+		fbdiv *= 2;
+		if (fbdiv >= 1024) {
+			madera_fll_err(fll, "FBDIV (%u) >= 1024\n", fbdiv);
+			return -EINVAL;
+		}
+	}
+
+	madera_fll_dbg(fll, "lockdet=%d, hp=0x%x, fbdiv:%d\n",
+		       lockdet_thr, hp, fbdiv);
+
+	/* Calculate N.K values */
+	fllgcd = gcd(fout, fbdiv * fref);
+	num = fout / fllgcd;
+	lambda = (fref * fbdiv) / fllgcd;
+	fll_n = num / lambda;
+	theta = num % lambda;
+
+	madera_fll_dbg(fll, "fll_n=%d, gcd=%d, theta=%d, lambda=%d\n",
+		       fll_n, fllgcd, theta, lambda);
+
+	/* Some sanity checks before any registers are written. */
+	if (fll_n < min_n || fll_n > max_n) {
+		madera_fll_err(fll, "N not in valid %s mode range %d-%d: %d\n",
+			       frac ? "fractional" : "integer", min_n, max_n,
+			       fll_n);
+		return -EINVAL;
+	}
+	if (fbdiv < 1 || (frac && fbdiv >= 1024) || (!frac && fbdiv >= 256)) {
+		madera_fll_err(fll, "Invalid fbdiv for %s mode (%u)\n",
+			       frac ? "fractional" : "integer", fbdiv);
+		return -EINVAL;
+	}
+
+	/* clear the ctrl_upd bit to guarantee we write to it later. */
+	regmap_write(madera->regmap,
+		     fll->base + MADERA_FLL_CONTROL_2_OFFS,
+		     fll_n << MADERA_FLL1_N_SHIFT);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_3_OFFS,
+			   MADERA_FLL1_THETA_MASK,
+			   theta << MADERA_FLL1_THETA_SHIFT);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_4_OFFS,
+			   MADERA_FLL1_LAMBDA_MASK,
+			   lambda << MADERA_FLL1_LAMBDA_SHIFT);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_5_OFFS,
+			   MADERA_FLL1_FB_DIV_MASK,
+			   fbdiv << MADERA_FLL1_FB_DIV_SHIFT);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_6_OFFS,
+			   MADERA_FLL1_REFCLK_DIV_MASK,
+			   refdiv << MADERA_FLL1_REFCLK_DIV_SHIFT);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_GAIN_OFFS,
+			   0xffff,
+			   gains);
+	val = hp << MADERA_FLL1_HP_SHIFT;
+	val |= 1 << MADERA_FLL1_PHASEDET_ENA_SHIFT;
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_10_OFFS,
+			   MADERA_FLL1_HP_MASK | MADERA_FLL1_PHASEDET_ENA_MASK,
+			   val);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_11_OFFS,
+			   MADERA_FLL1_LOCKDET_THR_MASK,
+			   lockdet_thr << MADERA_FLL1_LOCKDET_THR_SHIFT);
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL1_DIGITAL_TEST_1_OFFS,
+			   MADERA_FLL1_SYNC_EFS_ENA_MASK |
+			   MADERA_FLL1_CLK_VCO_FAST_SRC_MASK,
+			   fast_clk);
+
+	return 0;
+}
+
+static int madera_fllhj_enable(struct madera_fll *fll)
+{
+	struct madera *madera = fll->madera;
+	int already_enabled = madera_is_enabled_fll(fll, fll->base);
+	int ret;
+
+	if (already_enabled < 0)
+		return already_enabled;
+
+	if (!already_enabled)
+		pm_runtime_get_sync(madera->dev);
+
+	madera_fll_dbg(fll, "Enabling FLL, initially %s\n",
+		       already_enabled ? "enabled" : "disabled");
+
+	/* FLLn_HOLD must be set before configuring any registers */
+	regmap_update_bits(fll->madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_1_OFFS,
+			   MADERA_FLL1_HOLD_MASK,
+			   MADERA_FLL1_HOLD_MASK);
+
+	/* Apply refclk */
+	ret = madera_fllhj_apply(fll, fll->ref_freq);
+	if (ret) {
+		madera_fll_err(fll, "Failed to set FLL: %d\n", ret);
+		goto out;
+	}
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_1_OFFS,
+			   CS47L92_FLL1_REFCLK_SRC_MASK,
+			   fll->ref_src << CS47L92_FLL1_REFCLK_SRC_SHIFT);
+
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_1_OFFS,
+			   MADERA_FLL1_ENA_MASK,
+			   MADERA_FLL1_ENA_MASK);
+
+out:
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_11_OFFS,
+			   MADERA_FLL1_LOCKDET_MASK,
+			   MADERA_FLL1_LOCKDET_MASK);
+
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_2_OFFS,
+			   MADERA_FLL1_CTRL_UPD_MASK,
+			   MADERA_FLL1_CTRL_UPD_MASK);
+
+	/* Release the hold so that flln locks to external frequency */
+	regmap_update_bits(madera->regmap,
+			   fll->base + MADERA_FLL_CONTROL_1_OFFS,
+			   MADERA_FLL1_HOLD_MASK,
+			   0);
+
+	if (!already_enabled)
+		madera_wait_for_fll(fll, true);
+
+	return 0;
+}
+
+static int madera_fllhj_validate(struct madera_fll *fll,
+				 unsigned int ref_in,
+				 unsigned int fout)
+{
+	if (fout && !ref_in) {
+		madera_fll_err(fll, "fllout set without valid input clk\n");
+		return -EINVAL;
+	}
+
+	if (fll->fout && fout != fll->fout) {
+		madera_fll_err(fll, "Can't change output on active FLL\n");
+		return -EINVAL;
+	}
+
+	if (ref_in / MADERA_FLL_MAX_REFDIV > MADERA_FLLHJ_MAX_THRESH) {
+		madera_fll_err(fll, "Can't scale %dMHz to <=13MHz\n", ref_in);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+int madera_fllhj_set_refclk(struct madera_fll *fll, int source,
+			    unsigned int fin, unsigned int fout)
+{
+	int ret = 0;
+
+	/* To remain consistent with previous FLLs, we expect fout to be
+	 * provided in the form of the required sysclk rate, which is
+	 * 2x the calculated fll out.
+	 */
+	if (fout)
+		fout /= 2;
+
+	if (fll->ref_src == source && fll->ref_freq == fin &&
+	    fll->fout == fout)
+		return 0;
+
+	if (fin && fout && madera_fllhj_validate(fll, fin, fout))
+		return -EINVAL;
+
+	fll->ref_src = source;
+	fll->ref_freq = fin;
+	fll->fout = fout;
+
+	if (fout)
+		ret = madera_fllhj_enable(fll);
+	else
+		madera_fllhj_disable(fll);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(madera_fllhj_set_refclk);
+
 /**
  * madera_set_output_mode - Set the mode of the specified output
  *
diff --git a/sound/soc/codecs/madera.h b/sound/soc/codecs/madera.h
index 0af66f280770..1f3e8e230cf2 100644
--- a/sound/soc/codecs/madera.h
+++ b/sound/soc/codecs/madera.h
@@ -47,6 +47,7 @@
 #define MADERA_CLK_SYSCLK_3		6
 #define MADERA_CLK_ASYNCCLK_2		7
 #define MADERA_CLK_DSPCLK		8
+#define MADERA_CLK_OUTCLK		9
 
 #define MADERA_CLK_SRC_MCLK1		0x0
 #define MADERA_CLK_SRC_MCLK2		0x1
@@ -61,6 +62,12 @@
 #define MADERA_CLK_SRC_AIF4BCLK		0xB
 #define MADERA_CLK_SRC_FLLAO		0xF
 
+#define MADERA_OUTCLK_SYSCLK		0
+#define MADERA_OUTCLK_ASYNCCLK		1
+#define MADERA_OUTCLK_MCLK1		4
+#define MADERA_OUTCLK_MCLK2		5
+#define MADERA_OUTCLK_MCLK3		6
+
 #define MADERA_MIXER_VOL_MASK		0x00FE
 #define MADERA_MIXER_VOL_SHIFT		1
 #define MADERA_MIXER_VOL_WIDTH		7
@@ -326,6 +333,7 @@ extern const struct soc_enum madera_sample_rate[];
 extern const struct soc_enum madera_isrc_fsl[];
 extern const struct soc_enum madera_isrc_fsh[];
 extern const struct soc_enum madera_asrc1_rate[];
+extern const struct soc_enum madera_asrc1_bidir_rate[];
 extern const struct soc_enum madera_asrc2_rate[];
 extern const struct soc_enum madera_dfc_width[];
 extern const struct soc_enum madera_dfc_type[];
@@ -403,6 +411,8 @@ int madera_set_fll_syncclk(struct madera_fll *fll, int source,
 			   unsigned int fref, unsigned int fout);
 int madera_set_fll_ao_refclk(struct madera_fll *fll, int source,
 			     unsigned int fin, unsigned int fout);
+int madera_fllhj_set_refclk(struct madera_fll *fll, int source,
+			    unsigned int fin, unsigned int fout);
 
 int madera_core_init(struct madera_priv *priv);
 int madera_core_free(struct madera_priv *priv);
-- 
cgit v1.2.3-59-g8ed1b


From 68c907f10cd816cad2287167a1a1d77914a6d466 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:44 +0200
Subject: ASoC: codec2codec: run callbacks in order

When handling dai_link events on codec to codec links, run all .startup()
callbacks on sinks and sources before running any .hw_params(). Same goes
for hw_free() and shutdown(). This is closer to the behavior of regular
dai links

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 36 +++++++++++++++++++++++++++---------
 1 file changed, 27 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1d04612601ad..034b31fd2ecb 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3835,11 +3835,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				goto out;
 			}
 			source->active++;
-			ret = snd_soc_dai_hw_params(source, &substream, params);
-			if (ret < 0)
-				goto out;
-
-			dapm_update_dai_unlocked(&substream, params, source);
 		}
 
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
@@ -3853,6 +3848,23 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				goto out;
 			}
 			sink->active++;
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
+
+			ret = snd_soc_dai_hw_params(source, &substream, params);
+			if (ret < 0)
+				goto out;
+
+			dapm_update_dai_unlocked(&substream, params, source);
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
 			ret = snd_soc_dai_hw_params(sink, &substream, params);
 			if (ret < 0)
 				goto out;
@@ -3889,9 +3901,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
-
 			snd_soc_dai_hw_free(source, &substream);
+		}
+
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+			snd_soc_dai_hw_free(sink, &substream);
+		}
 
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
 			source->active--;
 			snd_soc_dai_shutdown(source, &substream);
 		}
@@ -3899,9 +3920,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
-
-			snd_soc_dai_hw_free(sink, &substream);
-
 			sink->active--;
 			snd_soc_dai_shutdown(sink, &substream);
 		}
-- 
cgit v1.2.3-59-g8ed1b


From 054d65004c6a008dfefbdae4fc1b46a3ad4e94c1 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:45 +0200
Subject: ASoC: codec2codec: name link using stream direction

At the moment, codec to codec dai link widgets are named after the
cpu dai and the 1st codec valid on the link. This might be confusing
if there is multiple valid codecs on the link for one stream
direction.

Instead, use the dai link name and the stream direction to name the
the dai link widget

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 12 ++++--------
 1 file changed, 4 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 034b31fd2ecb..7db4abd9a0a5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4056,8 +4056,7 @@ outfree_w_param:
 
 static struct snd_soc_dapm_widget *
 snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
-		     struct snd_soc_dapm_widget *source,
-		     struct snd_soc_dapm_widget *sink)
+		     char *id)
 {
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
@@ -4067,7 +4066,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 	int ret;
 
 	link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s",
-				   source->name, sink->name);
+				   rtd->dai_link->name, id);
 	if (!link_name)
 		return ERR_PTR(-ENOMEM);
 
@@ -4247,15 +4246,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 	}
 
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-
 		/* connect BE DAI playback if widgets are valid */
 		codec = codec_dai->playback_widget;
 
 		if (playback_cpu && codec) {
 			if (!playback) {
 				playback = snd_soc_dapm_new_dai(card, rtd,
-								playback_cpu,
-								codec);
+								"playback");
 				if (IS_ERR(playback)) {
 					dev_err(rtd->dev,
 						"ASoC: Failed to create DAI %s: %ld\n",
@@ -4284,8 +4281,7 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 		if (codec && capture_cpu) {
 			if (!capture) {
 				capture = snd_soc_dapm_new_dai(card, rtd,
-							       codec,
-							       capture_cpu);
+							       "capture");
 				if (IS_ERR(capture)) {
 					dev_err(rtd->dev,
 						"ASoC: Failed to create DAI %s: %ld\n",
-- 
cgit v1.2.3-59-g8ed1b


From 3dcfb397dad2ad55bf50de3c5d5a57090d35a18a Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:46 +0200
Subject: ASoC: codec2codec: deal with params when necessary

When there is an event on codec to codec dai_link, we only need to deal
with params if the event is SND_SOC_DAPM_PRE_PMU, when .hw_params() is
called. For the other events, it is useless.

Also, dealing with the codec to codec params just before calling
.hw_params() callbacks give change to either party on the link to alter
params content in .startup(), which might be useful in some cases

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 159 +++++++++++++++++++++++++++++----------------------
 1 file changed, 92 insertions(+), 67 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7db4abd9a0a5..6dcaf9ff6eb5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3764,25 +3764,59 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 
-static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
-				  struct snd_kcontrol *kcontrol, int event)
+static int
+snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
+			       struct snd_pcm_substream *substream)
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
-	struct snd_soc_pcm_runtime *rtd = w->priv;
-	const struct snd_soc_pcm_stream *config;
-	struct snd_pcm_substream substream;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_hw_params *params = NULL;
-	struct snd_pcm_runtime *runtime = NULL;
+	const struct snd_soc_pcm_stream *config = NULL;
 	unsigned int fmt;
-	int ret = 0;
+	int ret;
 
-	config = rtd->dai_link->params + rtd->params_select;
+	params = kzalloc(sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
 
-	if (WARN_ON(!config) ||
-	    WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
-		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
-		return -EINVAL;
+	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_soc_dapm_widget_for_each_source_path(w, path) {
+		source = path->source->priv;
+
+		ret = snd_soc_dai_startup(source, substream);
+		if (ret < 0) {
+			dev_err(source->dev,
+				"ASoC: startup() failed: %d\n", ret);
+			goto out;
+		}
+		source->active++;
+	}
+
+	substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_soc_dapm_widget_for_each_sink_path(w, path) {
+		sink = path->sink->priv;
+
+		ret = snd_soc_dai_startup(sink, substream);
+		if (ret < 0) {
+			dev_err(sink->dev,
+				"ASoC: startup() failed: %d\n", ret);
+			goto out;
+		}
+		sink->active++;
+	}
+
+	/*
+	 * Note: getting the config after .startup() gives a chance to
+	 * either party on the link to alter the configuration if
+	 * necessary
+	 */
+	config = rtd->dai_link->params + rtd->params_select;
+	if (WARN_ON(!config)) {
+		dev_err(w->dapm->dev, "ASoC: link config missing\n");
+		ret = -EINVAL;
+		goto out;
+	}
 
 	/* Be a little careful as we don't want to overflow the mask array */
 	if (config->formats) {
@@ -3790,27 +3824,62 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	} else {
 		dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n",
 			 config->formats);
-		fmt = 0;
-	}
 
-	/* Currently very limited parameter selection */
-	params = kzalloc(sizeof(*params), GFP_KERNEL);
-	if (!params) {
-		ret = -ENOMEM;
+		ret = -EINVAL;
 		goto out;
 	}
-	snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 
+	snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min =
 		config->rate_min;
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max =
 		config->rate_max;
-
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min
 		= config->channels_min;
 	hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max
 		= config->channels_max;
 
+	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_soc_dapm_widget_for_each_source_path(w, path) {
+		source = path->source->priv;
+
+		ret = snd_soc_dai_hw_params(source, substream, params);
+		if (ret < 0)
+			goto out;
+
+		dapm_update_dai_unlocked(substream, params, source);
+	}
+
+	substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_soc_dapm_widget_for_each_sink_path(w, path) {
+		sink = path->sink->priv;
+
+		ret = snd_soc_dai_hw_params(sink, substream, params);
+		if (ret < 0)
+			goto out;
+
+		dapm_update_dai_unlocked(substream, params, sink);
+	}
+
+out:
+	kfree(params);
+	return 0;
+}
+
+static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
+				  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_dapm_path *path;
+	struct snd_soc_dai *source, *sink;
+	struct snd_soc_pcm_runtime *rtd = w->priv;
+	struct snd_pcm_substream substream;
+	struct snd_pcm_runtime *runtime = NULL;
+	int ret = 0;
+
+	if (WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
+		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
+		return -EINVAL;
+
 	memset(&substream, 0, sizeof(substream));
 
 	/* Allocate a dummy snd_pcm_runtime for startup() and other ops() */
@@ -3824,53 +3893,10 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_soc_dapm_widget_for_each_source_path(w, path) {
-			source = path->source->priv;
-
-			ret = snd_soc_dai_startup(source, &substream);
-			if (ret < 0) {
-				dev_err(source->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
-			}
-			source->active++;
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_soc_dapm_widget_for_each_sink_path(w, path) {
-			sink = path->sink->priv;
-
-			ret = snd_soc_dai_startup(sink, &substream);
-			if (ret < 0) {
-				dev_err(sink->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
-			}
-			sink->active++;
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_soc_dapm_widget_for_each_source_path(w, path) {
-			source = path->source->priv;
-
-			ret = snd_soc_dai_hw_params(source, &substream, params);
-			if (ret < 0)
-				goto out;
-
-			dapm_update_dai_unlocked(&substream, params, source);
-		}
-
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_soc_dapm_widget_for_each_sink_path(w, path) {
-			sink = path->sink->priv;
-
-			ret = snd_soc_dai_hw_params(sink, &substream, params);
-			if (ret < 0)
-				goto out;
+		ret = snd_soc_dai_link_event_pre_pmu(w, &substream);
+		if (ret < 0)
+			goto out;
 
-			dapm_update_dai_unlocked(&substream, params, sink);
-		}
 		break;
 
 	case SND_SOC_DAPM_POST_PMU:
@@ -3932,7 +3958,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 out:
 	kfree(runtime);
-	kfree(params);
 	return ret;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 2448c813b12cf1fb08da1d20bdb2ae583cda2d6f Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 26 Jul 2019 09:42:44 +0300
Subject: ASoC: ti: davinci-mcasp: Support for correct symmetric sample bits

Implement custom snd_pcm_hw_rule to filter the available formats for the
second stream to make it symmetric and allow only formats which require
the same amount of bits on the bus as the running stream.

A simple constraint is not working correctly because for example:
the first stream is started with S24_LE

If we place 24 as constraint for the SAMPLE_BITS then the second stream
can not use S24_LE as it is physically 32bits.

If we would place 32 as constraint (physical width) then S32_LE would have
been allowed, but S24_3LE is not.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190726064244.3762-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-mcasp.c | 50 ++++++++++++++++++++++++++++++++++++++++----
 1 file changed, 46 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 87ae19329e71..7aa3c32e4a49 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -109,6 +109,7 @@ struct davinci_mcasp {
 
 	/* Used for comstraint setting on the second stream */
 	u32	channels;
+	int	max_format_width;
 	u8	active_serializers[2];
 
 #ifdef CONFIG_GPIOLIB
@@ -766,6 +767,8 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
 
 	if (mcasp->slot_width)
 		slot_width = mcasp->slot_width;
+	else if (mcasp->max_format_width)
+		slot_width = mcasp->max_format_width;
 	else
 		slot_width = sample_width;
 	/*
@@ -1233,8 +1236,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 
 	davinci_config_channel_size(mcasp, word_length);
 
-	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE)
+	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) {
 		mcasp->channels = channels;
+		if (!mcasp->max_format_width)
+			mcasp->max_format_width = word_length;
+	}
 
 	return 0;
 }
@@ -1286,6 +1292,28 @@ static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
 	return snd_mask_refine(fmt, &nfmt);
 }
 
+static int davinci_mcasp_hw_rule_format_width(struct snd_pcm_hw_params *params,
+					      struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, format_width;
+
+	snd_mask_none(&nfmt);
+	format_width = rd->mcasp->max_format_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) == format_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
 static const unsigned int davinci_mcasp_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000,
@@ -1463,7 +1491,20 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				   &mcasp->chconstr[substream->stream]);
 
-	if (mcasp->slot_width) {
+	if (mcasp->max_format_width) {
+		/*
+		 * Only allow formats which require same amount of bits on the
+		 * bus as the currently running stream
+		 */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_format_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
+	else if (mcasp->slot_width) {
 		/* Only allow formats require <= slot_width bits on the bus */
 		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_FORMAT,
@@ -1514,8 +1555,10 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream,
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
 		return;
 
-	if (!cpu_dai->active)
+	if (!cpu_dai->active) {
 		mcasp->channels = 0;
+		mcasp->max_format_width = 0;
+	}
 }
 
 static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
@@ -1571,7 +1614,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 		},
 		.ops 		= &davinci_mcasp_dai_ops,
 
-		.symmetric_samplebits	= 1,
 		.symmetric_rates	= 1,
 	},
 	{
-- 
cgit v1.2.3-59-g8ed1b


From dd65f7e19c6961ba6a69f7c925021b7a270cb950 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 26 Jul 2019 11:42:34 +0200
Subject: ALSA: hda - Show the fatal CORB/RIRB error more clearly

The last fallback of CORB/RIRB communication error recovery is to turn
on the single command mode, and this last resort usually means that
something is really screwed up.  Instead of a normal dev_err(), show
the error more clearly with dev_WARN() with the caller stack trace.

Also, show the bus-reset fallback also as an error, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_controller.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index c8d1b4316245..4b4cb0fd5102 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -864,10 +864,13 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
 	 */
 	if (hbus->allow_bus_reset && !hbus->response_reset && !hbus->in_reset) {
 		hbus->response_reset = 1;
+		dev_err(chip->card->dev,
+			"No response from codec, resetting bus: last cmd=0x%08x\n",
+			bus->last_cmd[addr]);
 		return -EAGAIN; /* give a chance to retry */
 	}
 
-	dev_err(chip->card->dev,
+	dev_WARN(chip->card->dev,
 		"azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n",
 		bus->last_cmd[addr]);
 	chip->single_cmd = 1;
-- 
cgit v1.2.3-59-g8ed1b


From 2127c01b7f63b06a21559f56a8c81a3c6535bd1a Mon Sep 17 00:00:00 2001
From: Jia-Ju Bai <baijiaju1990@gmail.com>
Date: Fri, 26 Jul 2019 10:14:42 +0800
Subject: ALSA: i2c: ak4xxx-adda: Fix a possible null pointer dereference in
 build_adc_controls()

In build_adc_controls(), there is an if statement on line 773 to check
whether ak->adc_info is NULL:
    if (! ak->adc_info ||
        ! ak->adc_info[mixer_ch].switch_name)

When ak->adc_info is NULL, it is used on line 792:
    knew.name = ak->adc_info[mixer_ch].selector_name;

Thus, a possible null-pointer dereference may occur.

To fix this bug, referring to lines 773 and 774, ak->adc_info
and ak->adc_info[mixer_ch].selector_name are checked before being used.

This bug is found by a static analysis tool STCheck written by us.

Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/i2c/other/ak4xxx-adda.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index 5f59316f982a..7d15093844b9 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -775,11 +775,12 @@ static int build_adc_controls(struct snd_akm4xxx *ak)
 				return err;
 
 			memset(&knew, 0, sizeof(knew));
-			knew.name = ak->adc_info[mixer_ch].selector_name;
-			if (!knew.name) {
+			if (!ak->adc_info ||
+				!ak->adc_info[mixer_ch].selector_name) {
 				knew.name = "Capture Channel";
 				knew.index = mixer_ch + ak->idx_offset * 2;
-			}
+			} else
+				knew.name = ak->adc_info[mixer_ch].selector_name;
 
 			knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 			knew.info = ak4xxx_capture_source_info;
-- 
cgit v1.2.3-59-g8ed1b


From c8415833ec242b9ddf73bf9e1057e12f9b0fcd16 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Fri, 26 Jul 2019 13:33:27 +0100
Subject: ASoC: codec2codec: fix missing return of error return code

Currently in function snd_soc_dai_link_event_pre_pmu the error return
code in variable err is being set but this is not actually being returned,
the function just returns zero even when there are failures. Fix this by
returning the error return code.

Addresses-Coverity: ("Unused value")
Fixes: 3dcfb397dad2 ("ASoC: codec2codec: deal with params when necessary")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190726123327.10467-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index be9bb05b0165..2d183e2d23de 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3776,7 +3776,7 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	struct snd_pcm_hw_params *params = NULL;
 	const struct snd_soc_pcm_stream *config = NULL;
 	unsigned int fmt;
-	int ret;
+	int ret = 0;
 
 	params = kzalloc(sizeof(*params), GFP_KERNEL);
 	if (!params)
@@ -3865,7 +3865,7 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 
 out:
 	kfree(params);
-	return 0;
+	return ret;
 }
 
 static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
-- 
cgit v1.2.3-59-g8ed1b


From 096769ea95380ef3048765aef676e8947bc368d4 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Fri, 26 Jul 2019 11:09:29 +0200
Subject: ASoC: Intel: Skylake: Remove static table index when parsing topology
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Currently when we remove and reload driver we use previous ref_count
value to start iterating over skl->modules which leads to out of table
access. To fix this just inline the function and calculate indexes
everytime we parse UUID token.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190726090929.27946-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 34 ++++++++++++----------------------
 1 file changed, 12 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 118866cd5075..c1c37ce759bd 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -3333,25 +3333,6 @@ static int skl_tplg_get_int_tkn(struct device *dev,
 	return tkn_count;
 }
 
-static int skl_tplg_get_manifest_uuid(struct device *dev,
-				struct skl_dev *skl,
-				struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn)
-{
-	static int ref_count;
-	struct skl_module *mod;
-
-	if (uuid_tkn->token == SKL_TKN_UUID) {
-		mod = skl->modules[ref_count];
-		guid_copy(&mod->uuid, (guid_t *)&uuid_tkn->uuid);
-		ref_count++;
-	} else {
-		dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token);
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
 /*
  * Fill the manifest structure by parsing the tokens based on the
  * type.
@@ -3362,6 +3343,7 @@ static int skl_tplg_get_manifest_tkn(struct device *dev,
 {
 	int tkn_count = 0, ret;
 	int off = 0, tuple_size = 0;
+	u8 uuid_index = 0;
 	struct snd_soc_tplg_vendor_array *array;
 	struct snd_soc_tplg_vendor_value_elem *tkn_elem;
 
@@ -3384,9 +3366,17 @@ static int skl_tplg_get_manifest_tkn(struct device *dev,
 			continue;
 
 		case SND_SOC_TPLG_TUPLE_TYPE_UUID:
-			ret = skl_tplg_get_manifest_uuid(dev, skl, array->uuid);
-			if (ret < 0)
-				return ret;
+			if (array->uuid->token != SKL_TKN_UUID) {
+				dev_err(dev, "Not an UUID token: %d\n",
+					array->uuid->token);
+				return -EINVAL;
+			}
+			if (uuid_index >= skl->nr_modules) {
+				dev_err(dev, "Too many UUID tokens\n");
+				return -EINVAL;
+			}
+			guid_copy(&skl->modules[uuid_index++]->uuid,
+				  (guid_t *)&array->uuid->uuid);
 
 			tuple_size += sizeof(*array->uuid);
 			continue;
-- 
cgit v1.2.3-59-g8ed1b


From 9e4d5c1be21f0c00e747e92186784f3298309b3e Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
Date: Mon, 29 Jul 2019 00:42:45 +0930
Subject: ALSA: usb-audio: Scarlett Gen 2 mixer interface

Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2
audio interfaces. Although the interfaces are USB compliant,
additional input/output level controls and hardware routing/mixing
functionality are available using proprietary USB requests.

Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/Makefile              |    1 +
 sound/usb/mixer.c               |    4 +
 sound/usb/mixer.h               |    4 +
 sound/usb/mixer_quirks.c        |    7 +
 sound/usb/mixer_scarlett_gen2.c | 2075 +++++++++++++++++++++++++++++++++++++++
 sound/usb/mixer_scarlett_gen2.h |    7 +
 6 files changed, 2098 insertions(+)
 create mode 100644 sound/usb/mixer_scarlett_gen2.c
 create mode 100644 sound/usb/mixer_scarlett_gen2.h

(limited to 'sound')

diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index e1ce257ab705..f4b7384caa9d 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -11,6 +11,7 @@ snd-usb-audio-objs := 	card.o \
 			mixer.o \
 			mixer_quirks.o \
 			mixer_scarlett.o \
+			mixer_scarlett_gen2.o \
 			mixer_us16x08.o \
 			pcm.o \
 			power.o \
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7498b5191b68..7b51ab4c35b0 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -3509,6 +3509,8 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer)
 		usb_kill_urb(mixer->urb);
 	if (mixer->rc_urb)
 		usb_kill_urb(mixer->rc_urb);
+	if (mixer->private_free)
+		mixer->private_free(mixer);
 	mixer->disconnected = true;
 }
 
@@ -3536,6 +3538,8 @@ static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
 int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer)
 {
 	snd_usb_mixer_inactivate(mixer);
+	if (mixer->private_suspend)
+		mixer->private_suspend(mixer);
 	return 0;
 }
 
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 394cd9107507..37e1b234c802 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -28,6 +28,10 @@ struct usb_mixer_interface {
 	struct media_mixer_ctl *media_mixer_ctl;
 
 	bool disconnected;
+
+	void *private_data;
+	void (*private_free)(struct usb_mixer_interface *mixer);
+	void (*private_suspend)(struct usb_mixer_interface *mixer);
 };
 
 #define MAX_CHANNELS	16	/* max logical channels */
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 199fa157a411..70805c15b0b5 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -32,6 +32,7 @@
 #include "mixer.h"
 #include "mixer_quirks.h"
 #include "mixer_scarlett.h"
+#include "mixer_scarlett_gen2.h"
 #include "mixer_us16x08.h"
 #include "helper.h"
 
@@ -2258,6 +2259,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 		err = snd_scarlett_controls_create(mixer);
 		break;
 
+	case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */
+	case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */
+	case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */
+		err = snd_scarlett_gen2_controls_create(mixer);
+		break;
+
 	case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */
 		err = snd_soundblaster_e1_switch_create(mixer);
 		break;
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
new file mode 100644
index 000000000000..7d460b1f1735
--- /dev/null
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -0,0 +1,2075 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *   Focusrite Scarlett 6i6/18i8/18i20 Gen 2 Driver for ALSA
+ *
+ *   Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu>
+ *
+ *   Based on the Scarlett (Gen 1) Driver for ALSA:
+ *
+ *   Copyright (c) 2013 by Tobias Hoffmann
+ *   Copyright (c) 2013 by Robin Gareus <robin at gareus.org>
+ *   Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de>
+ *   Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com>
+ *
+ *   Many codes borrowed from audio.c by
+ *     Alan Cox (alan at lxorguk.ukuu.org.uk)
+ *     Thomas Sailer (sailer at ife.ee.ethz.ch)
+ *
+ *   Code cleanup:
+ *   David Henningsson <david.henningsson at canonical.com>
+ */
+
+/* Mixer Interface for the Focusrite Scarlett 6i6/18i8/18i20 Gen 2 audio
+ * interface. Based on the Gen 1 driver and rewritten.
+ */
+
+/* The protocol was reverse engineered by looking at the communication
+ * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20
+ * (firmware 1083) using usbmon in July-August 2018.
+ *
+ * Scarlett 18i8 support added in April 2019.
+ *
+ * Scarlett 6i6 support added in June 2019 (thanks to Martin Wittmann
+ * for providing usbmon output and testing).
+ *
+ * This ALSA mixer gives access to:
+ *  - input, output, mixer-matrix muxes
+ *  - 18x10 mixer-matrix gain stages
+ *  - gain/volume controls
+ *  - level meters
+ *  - line/inst level and pad controls
+ *
+ * <ditaa>
+ *    /--------------\    18chn            20chn     /--------------\
+ *    | Hardware  in +--+------\    /-------------+--+ ALSA PCM out |
+ *    \--------------/  |      |    |             |  \--------------/
+ *                      |      |    |    /-----\  |
+ *                      |      |    |    |     |  |
+ *                      |      v    v    v     |  |
+ *                      |   +---------------+  |  |
+ *                      |    \ Matrix  Mux /   |  |
+ *                      |     +-----+-----+    |  |
+ *                      |           |          |  |
+ *                      |           |18chn     |  |
+ *                      |           |          |  |
+ *                      |           |     10chn|  |
+ *                      |           v          |  |
+ *                      |     +------------+   |  |
+ *                      |     | Mixer      |   |  |
+ *                      |     |     Matrix |   |  |
+ *                      |     |            |   |  |
+ *                      |     | 18x10 Gain |   |  |
+ *                      |     |   stages   |   |  |
+ *                      |     +-----+------+   |  |
+ *                      |           |          |  |
+ *                      |18chn      |10chn     |  |20chn
+ *                      |           |          |  |
+ *                      |           +----------/  |
+ *                      |           |             |
+ *                      v           v             v
+ *                      ===========================
+ *               +---------------+       +--—------------+
+ *                \ Output  Mux /         \ Capture Mux /
+ *                 +---+---+---+           +-----+-----+
+ *                     |   |                     |
+ *                10chn|   |                     |18chn
+ *                     |   |                     |
+ *  /--------------\   |   |                     |   /--------------\
+ *  | S/PDIF, ADAT |<--/   |10chn                \-->| ALSA PCM in  |
+ *  | Hardware out |       |                         \--------------/
+ *  \--------------/       |
+ *                         v
+ *                  +-------------+    Software gain per channel.
+ *                  | Master Gain |<-- 18i20 only: Switch per channel
+ *                  +------+------+    to select HW or SW gain control.
+ *                         |
+ *                         |10chn
+ *  /--------------\       |
+ *  | Analogue     |<------/
+ *  | Hardware out |
+ *  \--------------/
+ * </ditaa>
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/usb.h>
+#include <linux/moduleparam.h>
+
+#include <sound/control.h>
+#include <sound/tlv.h>
+
+#include "usbaudio.h"
+#include "mixer.h"
+#include "helper.h"
+
+#include "mixer_scarlett_gen2.h"
+
+/* device_setup value to enable */
+#define SCARLETT2_ENABLE 0x01
+
+/* some gui mixers can't handle negative ctl values */
+#define SCARLETT2_VOLUME_BIAS 127
+
+/* mixer range from -80dB to +6dB in 0.5dB steps */
+#define SCARLETT2_MIXER_MIN_DB -80
+#define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2)
+#define SCARLETT2_MIXER_MAX_DB 6
+#define SCARLETT2_MIXER_MAX_VALUE \
+	((SCARLETT2_MIXER_MAX_DB - SCARLETT2_MIXER_MIN_DB) * 2)
+
+/* map from (dB + 80) * 2 to mixer value
+ * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20)))
+ */
+static const u16 scarlett2_mixer_values[173] = {
+	0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2,
+	2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8,
+	9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21,
+	23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51,
+	54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115,
+	122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230,
+	244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460,
+	487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919,
+	973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634,
+	1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906,
+	3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168,
+	5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191,
+	9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430,
+	16345
+};
+
+/* Maximum number of analogue outputs */
+#define SCARLETT2_ANALOGUE_MAX 10
+
+/* Maximum number of level and pad switches */
+#define SCARLETT2_LEVEL_SWITCH_MAX 2
+#define SCARLETT2_PAD_SWITCH_MAX 4
+
+/* Maximum number of inputs to the mixer */
+#define SCARLETT2_INPUT_MIX_MAX 18
+
+/* Maximum number of outputs from the mixer */
+#define SCARLETT2_OUTPUT_MIX_MAX 10
+
+/* Maximum size of the data in the USB mux assignment message:
+ * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare
+ */
+#define SCARLETT2_MUX_MAX 64
+
+/* Number of meters:
+ * 18 inputs, 20 outputs, 18 matrix inputs
+ */
+#define SCARLETT2_NUM_METERS 56
+
+/* Hardware port types:
+ * - None (no input to mux)
+ * - Analogue I/O
+ * - S/PDIF I/O
+ * - ADAT I/O
+ * - Mixer I/O
+ * - PCM I/O
+ */
+enum {
+	SCARLETT2_PORT_TYPE_NONE = 0,
+	SCARLETT2_PORT_TYPE_ANALOGUE = 1,
+	SCARLETT2_PORT_TYPE_SPDIF = 2,
+	SCARLETT2_PORT_TYPE_ADAT = 3,
+	SCARLETT2_PORT_TYPE_MIX = 4,
+	SCARLETT2_PORT_TYPE_PCM = 5,
+	SCARLETT2_PORT_TYPE_COUNT = 6,
+};
+
+/* Count of total I/O and number available at each sample rate */
+enum {
+	SCARLETT2_PORT_IN = 0,
+	SCARLETT2_PORT_OUT = 1,
+	SCARLETT2_PORT_OUT_44 = 2,
+	SCARLETT2_PORT_OUT_88 = 3,
+	SCARLETT2_PORT_OUT_176 = 4,
+	SCARLETT2_PORT_DIRECTIONS = 5,
+};
+
+/* Hardware buttons on the 18i20 */
+#define SCARLETT2_BUTTON_MAX 2
+
+static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = {
+	"Mute", "Dim"
+};
+
+/* Description of each hardware port type:
+ * - id: hardware ID for this port type
+ * - num: number of sources/destinations of this port type
+ * - src_descr: printf format string for mux input selections
+ * - src_num_offset: added to channel number for the fprintf
+ * - dst_descr: printf format string for mixer controls
+ */
+struct scarlett2_ports {
+	u16 id;
+	int num[SCARLETT2_PORT_DIRECTIONS];
+	const char * const src_descr;
+	int src_num_offset;
+	const char * const dst_descr;
+};
+
+struct scarlett2_device_info {
+	u8 line_out_hw_vol; /* line out hw volume is sw controlled */
+	u8 button_count; /* number of buttons */
+	u8 level_input_count; /* inputs with level selectable */
+	u8 pad_input_count; /* inputs with pad selectable */
+	const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX];
+	struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT];
+};
+
+struct scarlett2_mixer_data {
+	struct usb_mixer_interface *mixer;
+	struct mutex usb_mutex; /* prevent sending concurrent USB requests */
+	struct mutex data_mutex; /* lock access to this data */
+	struct delayed_work work;
+	const struct scarlett2_device_info *info;
+	int num_mux_srcs;
+	u16 scarlett2_seq;
+	u8 vol_updated;
+	u8 master_vol;
+	u8 vol[SCARLETT2_ANALOGUE_MAX];
+	u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX];
+	u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX];
+	u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX];
+	u8 buttons[SCARLETT2_BUTTON_MAX];
+	struct snd_kcontrol *master_vol_ctl;
+	struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX];
+	struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX];
+	u8 mux[SCARLETT2_MUX_MAX];
+	u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX];
+};
+
+/*** Model-specific data ***/
+
+static const struct scarlett2_device_info s6i6_gen2_info = {
+	/* The first two analogue inputs can be switched between line
+	 * and instrument levels.
+	 */
+	.level_input_count = 2,
+
+	/* The first two analogue inputs have an optional pad. */
+	.pad_input_count = 2,
+
+	.line_out_descrs = {
+		"Monitor L",
+		"Monitor R",
+		"Headphones L",
+		"Headphones R",
+	},
+
+	.ports = {
+		{
+			.id = 0x000,
+			.num = { 1, 0, 8, 8, 8 },
+			.src_descr = "Off",
+			.src_num_offset = 0,
+		},
+		{
+			.id = 0x080,
+			.num = { 4, 4, 4, 4, 4 },
+			.src_descr = "Analogue %d",
+			.src_num_offset = 1,
+			.dst_descr = "Analogue Output %02d Playback"
+		},
+		{
+			.id = 0x180,
+			.num = { 2, 2, 2, 2, 2 },
+			.src_descr = "S/PDIF %d",
+			.src_num_offset = 1,
+			.dst_descr = "S/PDIF Output %d Playback"
+		},
+		{
+			.id = 0x300,
+			.num = { 10, 18, 18, 18, 18 },
+			.src_descr = "Mix %c",
+			.src_num_offset = 65,
+			.dst_descr = "Mixer Input %02d Capture"
+		},
+		{
+			.id = 0x600,
+			.num = { 6, 6, 6, 6, 6 },
+			.src_descr = "PCM %d",
+			.src_num_offset = 1,
+			.dst_descr = "PCM %02d Capture"
+		},
+	},
+};
+
+static const struct scarlett2_device_info s18i8_gen2_info = {
+	/* The first two analogue inputs can be switched between line
+	 * and instrument levels.
+	 */
+	.level_input_count = 2,
+
+	/* The first four analogue inputs have an optional pad. */
+	.pad_input_count = 4,
+
+	.line_out_descrs = {
+		"Monitor L",
+		"Monitor R",
+		"Headphones 1 L",
+		"Headphones 1 R",
+		"Headphones 2 L",
+		"Headphones 2 R",
+	},
+
+	.ports = {
+		{
+			.id = 0x000,
+			.num = { 1, 0, 8, 8, 4 },
+			.src_descr = "Off",
+			.src_num_offset = 0,
+		},
+		{
+			.id = 0x080,
+			.num = { 8, 6, 6, 6, 6 },
+			.src_descr = "Analogue %d",
+			.src_num_offset = 1,
+			.dst_descr = "Analogue Output %02d Playback"
+		},
+		{
+			/* S/PDIF outputs aren't available at 192KHz
+			 * but are included in the USB mux I/O
+			 * assignment message anyway
+			 */
+			.id = 0x180,
+			.num = { 2, 2, 2, 2, 2 },
+			.src_descr = "S/PDIF %d",
+			.src_num_offset = 1,
+			.dst_descr = "S/PDIF Output %d Playback"
+		},
+		{
+			.id = 0x200,
+			.num = { 8, 0, 0, 0, 0 },
+			.src_descr = "ADAT %d",
+			.src_num_offset = 1,
+		},
+		{
+			.id = 0x300,
+			.num = { 10, 18, 18, 18, 18 },
+			.src_descr = "Mix %c",
+			.src_num_offset = 65,
+			.dst_descr = "Mixer Input %02d Capture"
+		},
+		{
+			.id = 0x600,
+			.num = { 20, 18, 18, 14, 10 },
+			.src_descr = "PCM %d",
+			.src_num_offset = 1,
+			.dst_descr = "PCM %02d Capture"
+		},
+	},
+};
+
+static const struct scarlett2_device_info s18i20_gen2_info = {
+	/* The analogue line outputs on the 18i20 can be switched
+	 * between software and hardware volume control
+	 */
+	.line_out_hw_vol = 1,
+
+	/* Mute and dim buttons */
+	.button_count = 2,
+
+	.line_out_descrs = {
+		"Monitor L",
+		"Monitor R",
+		NULL,
+		NULL,
+		NULL,
+		NULL,
+		"Headphones 1 L",
+		"Headphones 1 R",
+		"Headphones 2 L",
+		"Headphones 2 R",
+	},
+
+	.ports = {
+		{
+			.id = 0x000,
+			.num = { 1, 0, 8, 8, 6 },
+			.src_descr = "Off",
+			.src_num_offset = 0,
+		},
+		{
+			.id = 0x080,
+			.num = { 8, 10, 10, 10, 10 },
+			.src_descr = "Analogue %d",
+			.src_num_offset = 1,
+			.dst_descr = "Analogue Output %02d Playback"
+		},
+		{
+			/* S/PDIF outputs aren't available at 192KHz
+			 * but are included in the USB mux I/O
+			 * assignment message anyway
+			 */
+			.id = 0x180,
+			.num = { 2, 2, 2, 2, 2 },
+			.src_descr = "S/PDIF %d",
+			.src_num_offset = 1,
+			.dst_descr = "S/PDIF Output %d Playback"
+		},
+		{
+			.id = 0x200,
+			.num = { 8, 8, 8, 4, 0 },
+			.src_descr = "ADAT %d",
+			.src_num_offset = 1,
+			.dst_descr = "ADAT Output %d Playback"
+		},
+		{
+			.id = 0x300,
+			.num = { 10, 18, 18, 18, 18 },
+			.src_descr = "Mix %c",
+			.src_num_offset = 65,
+			.dst_descr = "Mixer Input %02d Capture"
+		},
+		{
+			.id = 0x600,
+			.num = { 20, 18, 18, 14, 10 },
+			.src_descr = "PCM %d",
+			.src_num_offset = 1,
+			.dst_descr = "PCM %02d Capture"
+		},
+	},
+};
+
+/* get the starting port index number for a given port type/direction */
+static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports,
+					int direction, int port_type)
+{
+	int i, num = 0;
+
+	for (i = 0; i < port_type; i++)
+		num += ports[i].num[direction];
+
+	return num;
+}
+
+/*** USB Interactions ***/
+
+/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */
+#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5
+#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4
+#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64
+#define SCARLETT2_USB_INTERRUPT_INTERVAL 3
+
+/* Interrupt flags for volume and mute/dim button changes */
+#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000
+#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000
+
+/* Commands for sending/receiving requests/responses */
+#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2
+#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3
+
+#define SCARLETT2_USB_INIT_SEQ 0x00000000
+#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001
+#define SCARLETT2_USB_SET_MIX 0x00002002
+#define SCARLETT2_USB_SET_MUX 0x00003002
+#define SCARLETT2_USB_GET_DATA 0x00800000
+#define SCARLETT2_USB_SET_DATA 0x00800001
+#define SCARLETT2_USB_DATA_CMD 0x00800002
+#define SCARLETT2_USB_CONFIG_SAVE 6
+
+#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31
+#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1
+
+/* volume status is read together (matches scarlett2_config_items[]) */
+struct scarlett2_usb_volume_status {
+	/* mute & dim buttons */
+	u8 buttons[SCARLETT2_BUTTON_MAX];
+
+	u8 pad1;
+
+	/* software volume setting */
+	s16 sw_vol[SCARLETT2_ANALOGUE_MAX];
+
+	/* actual volume of output inc. dim (-18dB) */
+	s16 hw_vol[SCARLETT2_ANALOGUE_MAX];
+
+	u8 pad2[SCARLETT2_ANALOGUE_MAX];
+
+	/* sw (0) or hw (1) controlled */
+	u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX];
+
+	u8 pad3[6];
+
+	/* front panel volume knob */
+	s16 master_vol;
+} __packed;
+
+/* Configuration parameters that can be read and written */
+enum {
+	SCARLETT2_CONFIG_BUTTONS = 0,
+	SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1,
+	SCARLETT2_CONFIG_SW_HW_SWITCH = 2,
+	SCARLETT2_CONFIG_LEVEL_SWITCH = 3,
+	SCARLETT2_CONFIG_PAD_SWITCH = 4,
+	SCARLETT2_CONFIG_COUNT = 5
+};
+
+/* Location, size, and activation command number for the configuration
+ * parameters
+ */
+struct scarlett2_config {
+	u8 offset;
+	u8 size;
+	u8 activate;
+};
+
+static const struct scarlett2_config
+		scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = {
+	/* Mute/Dim Buttons */
+	{
+		.offset = 0x31,
+		.size = 1,
+		.activate = 2
+	},
+
+	/* Line Out Volume */
+	{
+		.offset = 0x34,
+		.size = 2,
+		.activate = 1
+	},
+
+	/* SW/HW Volume Switch */
+	{
+		.offset = 0x66,
+		.size = 1,
+		.activate = 3
+	},
+
+	/* Level Switch */
+	{
+		.offset = 0x7c,
+		.size = 1,
+		.activate = 7
+	},
+
+	/* Pad Switch */
+	{
+		.offset = 0x84,
+		.size = 1,
+		.activate = 8
+	}
+};
+
+/* proprietary request/response format */
+struct scarlett2_usb_packet {
+	u32 cmd;
+	u16 size;
+	u16 seq;
+	u32 error;
+	u32 pad;
+	u8 data[];
+};
+
+#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet))
+
+static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private,
+					  struct scarlett2_usb_packet *req,
+					  u32 cmd, u16 req_size)
+{
+	/* sequence must go up by 1 for each request */
+	u16 seq = private->scarlett2_seq++;
+
+	req->cmd = cpu_to_le32(cmd);
+	req->size = cpu_to_le16(req_size);
+	req->seq = cpu_to_le16(seq);
+	req->error = 0;
+	req->pad = 0;
+}
+
+/* Send a proprietary format request to the Scarlett interface */
+static int scarlett2_usb(
+	struct usb_mixer_interface *mixer, u32 cmd,
+	void *req_data, u16 req_size, void *resp_data, u16 resp_size)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size;
+	u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size;
+	struct scarlett2_usb_packet *req = NULL, *resp = NULL;
+	int err = 0;
+
+	req = kmalloc(req_buf_size, GFP_KERNEL);
+	if (!req) {
+		err = -ENOMEM;
+		goto error;
+	}
+
+	resp = kmalloc(resp_buf_size, GFP_KERNEL);
+	if (!resp) {
+		err = -ENOMEM;
+		goto error;
+	}
+
+	mutex_lock(&private->usb_mutex);
+
+	/* build request message and send it */
+
+	scarlett2_fill_request_header(private, req, cmd, req_size);
+
+	if (req_size)
+		memcpy(req->data, req_data, req_size);
+
+	err = snd_usb_ctl_msg(mixer->chip->dev,
+			usb_sndctrlpipe(mixer->chip->dev, 0),
+			SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ,
+			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+			0,
+			SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE,
+			req,
+			req_buf_size);
+
+	if (err != req_buf_size) {
+		usb_audio_err(
+			mixer->chip,
+			"Scarlett Gen 2 USB request result cmd %x was %d\n",
+			cmd, err);
+		err = -EINVAL;
+		goto unlock;
+	}
+
+	/* send a second message to get the response */
+
+	err = snd_usb_ctl_msg(mixer->chip->dev,
+			usb_sndctrlpipe(mixer->chip->dev, 0),
+			SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP,
+			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+			0,
+			SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE,
+			resp,
+			resp_buf_size);
+
+	/* validate the response */
+
+	if (err != resp_buf_size) {
+		usb_audio_err(
+			mixer->chip,
+			"Scarlett Gen 2 USB response result cmd %x was %d\n",
+			cmd, err);
+		err = -EINVAL;
+		goto unlock;
+	}
+
+	if (resp->cmd != req->cmd ||
+	    resp->seq != req->seq ||
+	    resp_size != le16_to_cpu(resp->size) ||
+	    resp->error ||
+	    resp->pad) {
+		usb_audio_err(
+			mixer->chip,
+			"Scarlett Gen 2 USB invalid response; "
+			   "cmd tx/rx %d/%d seq %d/%d size %d/%d "
+			   "error %d pad %d\n",
+			le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd),
+			le16_to_cpu(req->seq), le16_to_cpu(resp->seq),
+			resp_size, le16_to_cpu(resp->size),
+			le16_to_cpu(resp->error),
+			le16_to_cpu(resp->pad));
+		err = -EINVAL;
+		goto unlock;
+	}
+
+	if (resp_size > 0)
+		memcpy(resp_data, resp->data, resp_size);
+
+unlock:
+	mutex_unlock(&private->usb_mutex);
+error:
+	kfree(req);
+	kfree(resp);
+	return err;
+}
+
+/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */
+static void scarlett2_config_save(struct usb_mixer_interface *mixer)
+{
+	u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE);
+
+	scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD,
+		      &req, sizeof(u32),
+		      NULL, 0);
+}
+
+/* Delayed work to save config */
+static void scarlett2_config_save_work(struct work_struct *work)
+{
+	struct scarlett2_mixer_data *private =
+		container_of(work, struct scarlett2_mixer_data, work.work);
+
+	scarlett2_config_save(private->mixer);
+}
+
+/* Send a USB message to set a configuration parameter (volume level,
+ * sw/hw volume switch, line/inst level switch, or pad switch)
+ */
+static int scarlett2_usb_set_config(
+	struct usb_mixer_interface *mixer,
+	int config_item_num, int index, int value)
+{
+	const struct scarlett2_config config_item =
+	       scarlett2_config_items[config_item_num];
+	struct {
+		u32 offset;
+		u32 bytes;
+		s32 value;
+	} __packed req;
+	u32 req2;
+	int err;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	/* Cancel any pending NVRAM save */
+	cancel_delayed_work_sync(&private->work);
+
+	/* Send the configuration parameter data */
+	req.offset = cpu_to_le32(config_item.offset + index * config_item.size);
+	req.bytes = cpu_to_le32(config_item.size);
+	req.value = cpu_to_le32(value);
+	err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA,
+			    &req, sizeof(u32) * 2 + config_item.size,
+			    NULL, 0);
+	if (err < 0)
+		return err;
+
+	/* Activate the change */
+	req2 = cpu_to_le32(config_item.activate);
+	err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD,
+			    &req2, sizeof(req2), NULL, 0);
+	if (err < 0)
+		return err;
+
+	/* Schedule the change to be written to NVRAM */
+	schedule_delayed_work(&private->work, msecs_to_jiffies(2000));
+
+	return 0;
+}
+
+/* Send a USB message to get data; result placed in *buf */
+static int scarlett2_usb_get(
+	struct usb_mixer_interface *mixer,
+	int offset, void *buf, int size)
+{
+	struct {
+		u32 offset;
+		u32 size;
+	} __packed req;
+
+	req.offset = cpu_to_le32(offset);
+	req.size = cpu_to_le32(size);
+	return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA,
+			     &req, sizeof(req), buf, size);
+}
+
+/* Send a USB message to get configuration parameters; result placed in *buf */
+static int scarlett2_usb_get_config(
+	struct usb_mixer_interface *mixer,
+	int config_item_num, int count, void *buf)
+{
+	const struct scarlett2_config config_item =
+	       scarlett2_config_items[config_item_num];
+	int size = config_item.size * count;
+
+	return scarlett2_usb_get(mixer, config_item.offset, buf, size);
+}
+
+/* Send a USB message to get volume status; result placed in *buf */
+static int scarlett2_usb_get_volume_status(
+	struct usb_mixer_interface *mixer,
+	struct scarlett2_usb_volume_status *buf)
+{
+	return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET,
+				 buf, sizeof(*buf));
+}
+
+/* Send a USB message to set the volumes for all inputs of one mix
+ * (values obtained from private->mix[])
+ */
+static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer,
+				     int mix_num)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_device_info *info = private->info;
+
+	struct {
+		u16 mix_num;
+		u16 data[SCARLETT2_INPUT_MIX_MAX];
+	} __packed req;
+
+	int i, j;
+	int num_mixer_in =
+		info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT];
+
+	req.mix_num = cpu_to_le16(mix_num);
+
+	for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++)
+		req.data[i] = cpu_to_le16(
+			scarlett2_mixer_values[private->mix[j]]
+		);
+
+	return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX,
+			     &req, (num_mixer_in + 1) * sizeof(u16),
+			     NULL, 0);
+}
+
+/* Convert a port number index (per info->ports) to a hardware ID */
+static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports,
+				       int num)
+{
+	int port_type;
+
+	for (port_type = 0;
+	     port_type < SCARLETT2_PORT_TYPE_COUNT;
+	     port_type++) {
+		if (num < ports[port_type].num[SCARLETT2_PORT_IN])
+			return ports[port_type].id | num;
+		num -= ports[port_type].num[SCARLETT2_PORT_IN];
+	}
+
+	/* Oops */
+	return 0;
+}
+
+/* Send USB messages to set mux inputs */
+static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_device_info *info = private->info;
+	const struct scarlett2_ports *ports = info->ports;
+	int rate, port_dir_rate;
+
+	static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = {
+		SCARLETT2_PORT_TYPE_PCM,
+		SCARLETT2_PORT_TYPE_ANALOGUE,
+		SCARLETT2_PORT_TYPE_SPDIF,
+		SCARLETT2_PORT_TYPE_ADAT,
+		SCARLETT2_PORT_TYPE_MIX,
+		SCARLETT2_PORT_TYPE_NONE,
+	};
+
+	struct {
+		u16 pad;
+		u16 num;
+		u32 data[SCARLETT2_MUX_MAX];
+	} __packed req;
+
+	req.pad = 0;
+
+	/* mux settings for each rate */
+	for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44;
+	     port_dir_rate <= SCARLETT2_PORT_OUT_176;
+	     rate++, port_dir_rate++) {
+		int order_num, i, err;
+
+		req.num = cpu_to_le16(rate);
+
+		for (order_num = 0, i = 0;
+		     order_num < SCARLETT2_PORT_TYPE_COUNT;
+		     order_num++) {
+			int port_type = assignment_order[order_num];
+			int j = scarlett2_get_port_start_num(ports,
+							     SCARLETT2_PORT_OUT,
+							     port_type);
+			int port_id = ports[port_type].id;
+			int channel;
+
+			for (channel = 0;
+			     channel < ports[port_type].num[port_dir_rate];
+			     channel++, i++, j++)
+				/* lower 12 bits for the destination and
+				 * next 12 bits for the source
+				 */
+				req.data[i] = !port_id
+					? 0
+					: cpu_to_le32(
+						port_id |
+						channel |
+						scarlett2_mux_src_num_to_id(
+							ports, private->mux[j]
+						) << 12
+					  );
+
+			/* skip private->mux[j] entries not output */
+			j += ports[port_type].num[SCARLETT2_PORT_OUT] -
+			     ports[port_type].num[port_dir_rate];
+		}
+
+		err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX,
+				    &req, (i + 1) * sizeof(u32),
+				    NULL, 0);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* Send USB message to get meter levels */
+static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer,
+					  u16 *levels)
+{
+	struct {
+		u16 pad;
+		u16 num_meters;
+		u32 magic;
+	} __packed req;
+	u32 resp[SCARLETT2_NUM_METERS];
+	int i, err;
+
+	req.pad = 0;
+	req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS);
+	req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC);
+	err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS,
+			    &req, sizeof(req), resp, sizeof(resp));
+	if (err < 0)
+		return err;
+
+	/* copy, convert to u16 */
+	for (i = 0; i < SCARLETT2_NUM_METERS; i++)
+		levels[i] = resp[i];
+
+	return 0;
+}
+
+/*** Control Functions ***/
+
+/* helper function to create a new control */
+static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer,
+				 const struct snd_kcontrol_new *ncontrol,
+				 int index, int channels, const char *name,
+				 struct snd_kcontrol **kctl_return)
+{
+	struct snd_kcontrol *kctl;
+	struct usb_mixer_elem_info *elem;
+	int err;
+
+	elem = kzalloc(sizeof(*elem), GFP_KERNEL);
+	if (!elem)
+		return -ENOMEM;
+
+	elem->head.mixer = mixer;
+	elem->control = index;
+	elem->head.id = index;
+	elem->channels = channels;
+
+	kctl = snd_ctl_new1(ncontrol, elem);
+	if (!kctl) {
+		kfree(elem);
+		return -ENOMEM;
+	}
+	kctl->private_free = snd_usb_mixer_elem_free;
+
+	strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
+
+	err = snd_usb_mixer_add_control(&elem->head, kctl);
+	if (err < 0)
+		return err;
+
+	if (kctl_return)
+		*kctl_return = kctl;
+
+	return 0;
+}
+
+/*** Analogue Line Out Volume Controls ***/
+
+/* Update hardware volume controls after receiving notification that
+ * they have changed
+ */
+static int scarlett2_update_volumes(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_ports *ports = private->info->ports;
+	struct scarlett2_usb_volume_status volume_status;
+	int num_line_out =
+		ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT];
+	int err, i;
+
+	private->vol_updated = 0;
+
+	err = scarlett2_usb_get_volume_status(mixer, &volume_status);
+	if (err < 0)
+		return err;
+
+	private->master_vol = clamp(
+		volume_status.master_vol + SCARLETT2_VOLUME_BIAS,
+		0, SCARLETT2_VOLUME_BIAS);
+
+	for (i = 0; i < num_line_out; i++) {
+		if (private->vol_sw_hw_switch[i])
+			private->vol[i] = private->master_vol;
+	}
+
+	for (i = 0; i < private->info->button_count; i++)
+		private->buttons[i] = !!volume_status.buttons[i];
+
+	return 0;
+}
+
+static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl,
+				     struct snd_ctl_elem_info *uinfo)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = elem->channels;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = SCARLETT2_VOLUME_BIAS;
+	uinfo->value.integer.step = 1;
+	return 0;
+}
+
+static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl,
+					   struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	if (private->vol_updated) {
+		mutex_lock(&private->data_mutex);
+		scarlett2_update_volumes(mixer);
+		mutex_unlock(&private->data_mutex);
+	}
+
+	ucontrol->value.integer.value[0] = private->master_vol;
+	return 0;
+}
+
+static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	int index = elem->control;
+
+	if (private->vol_updated) {
+		mutex_lock(&private->data_mutex);
+		scarlett2_update_volumes(mixer);
+		mutex_unlock(&private->data_mutex);
+	}
+
+	ucontrol->value.integer.value[0] = private->vol[index];
+	return 0;
+}
+
+static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	int index = elem->control;
+	int oval, val, err = 0;
+
+	mutex_lock(&private->data_mutex);
+
+	oval = private->vol[index];
+	val = ucontrol->value.integer.value[0];
+
+	if (oval == val)
+		goto unlock;
+
+	private->vol[index] = val;
+	err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME,
+				       index, val - SCARLETT2_VOLUME_BIAS);
+	if (err == 0)
+		err = 1;
+
+unlock:
+	mutex_unlock(&private->data_mutex);
+	return err;
+}
+
+static const DECLARE_TLV_DB_MINMAX(
+	db_scale_scarlett2_gain, -SCARLETT2_VOLUME_BIAS * 100, 0
+);
+
+static const struct snd_kcontrol_new scarlett2_master_volume_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = SNDRV_CTL_ELEM_ACCESS_READ |
+		  SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+	.name = "",
+	.info = scarlett2_volume_ctl_info,
+	.get  = scarlett2_master_volume_ctl_get,
+	.private_value = 0, /* max value */
+	.tlv = { .p = db_scale_scarlett2_gain }
+};
+
+static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		  SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+	.name = "",
+	.info = scarlett2_volume_ctl_info,
+	.get  = scarlett2_volume_ctl_get,
+	.put  = scarlett2_volume_ctl_put,
+	.private_value = 0, /* max value */
+	.tlv = { .p = db_scale_scarlett2_gain }
+};
+
+/*** HW/SW Volume Switch Controls ***/
+
+static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl,
+					 struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const values[2] = {
+		"SW", "HW"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, 2, values);
+}
+
+static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct scarlett2_mixer_data *private = elem->head.mixer->private_data;
+
+	ucontrol->value.enumerated.item[0] =
+		private->vol_sw_hw_switch[elem->control];
+	return 0;
+}
+
+static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	int index = elem->control;
+	int oval, val, err = 0;
+
+	mutex_lock(&private->data_mutex);
+
+	oval = private->vol_sw_hw_switch[index];
+	val = !!ucontrol->value.integer.value[0];
+
+	if (oval == val)
+		goto unlock;
+
+	private->vol_sw_hw_switch[index] = val;
+
+	/* Change access mode to RO (hardware controlled volume)
+	 * or RW (software controlled volume)
+	 */
+	if (val)
+		private->vol_ctls[index]->vd[0].access &=
+			~SNDRV_CTL_ELEM_ACCESS_WRITE;
+	else
+		private->vol_ctls[index]->vd[0].access |=
+			SNDRV_CTL_ELEM_ACCESS_WRITE;
+
+	/* Reset volume to master volume */
+	private->vol[index] = private->master_vol;
+
+	/* Set SW volume to current HW volume */
+	err = scarlett2_usb_set_config(
+		mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME,
+		index, private->master_vol - SCARLETT2_VOLUME_BIAS);
+	if (err < 0)
+		goto unlock;
+
+	/* Notify of RO/RW change */
+	snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO,
+		       &private->vol_ctls[index]->id);
+
+	/* Send SW/HW switch change to the device */
+	err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH,
+				       index, val);
+
+unlock:
+	mutex_unlock(&private->data_mutex);
+	return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "",
+	.info = scarlett2_sw_hw_enum_ctl_info,
+	.get  = scarlett2_sw_hw_enum_ctl_get,
+	.put  = scarlett2_sw_hw_enum_ctl_put,
+};
+
+/*** Line Level/Instrument Level Switch Controls ***/
+
+static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl,
+					 struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const values[2] = {
+		"Line", "Inst"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, 2, values);
+}
+
+static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct scarlett2_mixer_data *private = elem->head.mixer->private_data;
+
+	ucontrol->value.enumerated.item[0] =
+		private->level_switch[elem->control];
+	return 0;
+}
+
+static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	int index = elem->control;
+	int oval, val, err = 0;
+
+	mutex_lock(&private->data_mutex);
+
+	oval = private->level_switch[index];
+	val = !!ucontrol->value.integer.value[0];
+
+	if (oval == val)
+		goto unlock;
+
+	private->level_switch[index] = val;
+
+	/* Send switch change to the device */
+	err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH,
+				       index, val);
+
+unlock:
+	mutex_unlock(&private->data_mutex);
+	return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_level_enum_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "",
+	.info = scarlett2_level_enum_ctl_info,
+	.get  = scarlett2_level_enum_ctl_get,
+	.put  = scarlett2_level_enum_ctl_put,
+};
+
+/*** Pad Switch Controls ***/
+
+static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct scarlett2_mixer_data *private = elem->head.mixer->private_data;
+
+	ucontrol->value.enumerated.item[0] =
+		private->pad_switch[elem->control];
+	return 0;
+}
+
+static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	int index = elem->control;
+	int oval, val, err = 0;
+
+	mutex_lock(&private->data_mutex);
+
+	oval = private->pad_switch[index];
+	val = !!ucontrol->value.integer.value[0];
+
+	if (oval == val)
+		goto unlock;
+
+	private->pad_switch[index] = val;
+
+	/* Send switch change to the device */
+	err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH,
+				       index, val);
+
+unlock:
+	mutex_unlock(&private->data_mutex);
+	return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_pad_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "",
+	.info = snd_ctl_boolean_mono_info,
+	.get  = scarlett2_pad_ctl_get,
+	.put  = scarlett2_pad_ctl_put,
+};
+
+/*** Mute/Dim Controls ***/
+
+static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	if (private->vol_updated) {
+		mutex_lock(&private->data_mutex);
+		scarlett2_update_volumes(mixer);
+		mutex_unlock(&private->data_mutex);
+	}
+
+	ucontrol->value.enumerated.item[0] = private->buttons[elem->control];
+	return 0;
+}
+
+static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	int index = elem->control;
+	int oval, val, err = 0;
+
+	mutex_lock(&private->data_mutex);
+
+	oval = private->buttons[index];
+	val = !!ucontrol->value.integer.value[0];
+
+	if (oval == val)
+		goto unlock;
+
+	private->buttons[index] = val;
+
+	/* Send switch change to the device */
+	err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS,
+				       index, val);
+
+unlock:
+	mutex_unlock(&private->data_mutex);
+	return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_button_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "",
+	.info = snd_ctl_boolean_mono_info,
+	.get  = scarlett2_button_ctl_get,
+	.put  = scarlett2_button_ctl_put
+};
+
+/*** Create the analogue output controls ***/
+
+static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_device_info *info = private->info;
+	const struct scarlett2_ports *ports = info->ports;
+	int num_line_out =
+		ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT];
+	int err, i;
+	char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+	/* Add R/O HW volume control */
+	if (info->line_out_hw_vol) {
+		snprintf(s, sizeof(s), "Master HW Playback Volume");
+		err = scarlett2_add_new_ctl(mixer,
+					    &scarlett2_master_volume_ctl,
+					    0, 1, s, &private->master_vol_ctl);
+		if (err < 0)
+			return err;
+	}
+
+	/* Add volume controls */
+	for (i = 0; i < num_line_out; i++) {
+
+		/* Fader */
+		if (info->line_out_descrs[i])
+			snprintf(s, sizeof(s),
+				 "Line %02d (%s) Playback Volume",
+				 i + 1, info->line_out_descrs[i]);
+		else
+			snprintf(s, sizeof(s),
+				 "Line %02d Playback Volume",
+				 i + 1);
+		err = scarlett2_add_new_ctl(mixer,
+					    &scarlett2_line_out_volume_ctl,
+					    i, 1, s, &private->vol_ctls[i]);
+		if (err < 0)
+			return err;
+
+		/* Make the fader read-only if the SW/HW switch is set to HW */
+		if (private->vol_sw_hw_switch[i])
+			private->vol_ctls[i]->vd[0].access &=
+				~SNDRV_CTL_ELEM_ACCESS_WRITE;
+
+		/* SW/HW Switch */
+		if (info->line_out_hw_vol) {
+			snprintf(s, sizeof(s),
+				 "Line Out %02d Volume Control Playback Enum",
+				 i + 1);
+			err = scarlett2_add_new_ctl(mixer,
+						    &scarlett2_sw_hw_enum_ctl,
+						    i, 1, s, NULL);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	/* Add HW button controls */
+	for (i = 0; i < private->info->button_count; i++) {
+		err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl,
+					    i, 1, scarlett2_button_names[i],
+					    &private->button_ctls[i]);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/*** Create the analogue input controls ***/
+
+static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_device_info *info = private->info;
+	int err, i;
+	char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+	/* Add input level (line/inst) controls */
+	for (i = 0; i < info->level_input_count; i++) {
+		snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1);
+		err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl,
+					    i, 1, s, NULL);
+		if (err < 0)
+			return err;
+	}
+
+	/* Add input pad controls */
+	for (i = 0; i < info->pad_input_count; i++) {
+		snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1);
+		err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl,
+					    i, 1, s, NULL);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/*** Mixer Volume Controls ***/
+
+static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl,
+				    struct snd_ctl_elem_info *uinfo)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = elem->channels;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = SCARLETT2_MIXER_MAX_VALUE;
+	uinfo->value.integer.step = 1;
+	return 0;
+}
+
+static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct scarlett2_mixer_data *private = elem->head.mixer->private_data;
+
+	ucontrol->value.integer.value[0] = private->mix[elem->control];
+	return 0;
+}
+
+static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_device_info *info = private->info;
+	const struct scarlett2_ports *ports = info->ports;
+	int oval, val, num_mixer_in, mix_num, err = 0;
+
+	mutex_lock(&private->data_mutex);
+
+	oval = private->mix[elem->control];
+	val = ucontrol->value.integer.value[0];
+	num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT];
+	mix_num = elem->control / num_mixer_in;
+
+	if (oval == val)
+		goto unlock;
+
+	private->mix[elem->control] = val;
+	err = scarlett2_usb_set_mix(mixer, mix_num);
+	if (err == 0)
+		err = 1;
+
+unlock:
+	mutex_unlock(&private->data_mutex);
+	return err;
+}
+
+static const DECLARE_TLV_DB_MINMAX(
+	db_scale_scarlett2_mixer,
+	SCARLETT2_MIXER_MIN_DB * 100,
+	SCARLETT2_MIXER_MAX_DB * 100
+);
+
+static const struct snd_kcontrol_new scarlett2_mixer_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		  SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+	.name = "",
+	.info = scarlett2_mixer_ctl_info,
+	.get  = scarlett2_mixer_ctl_get,
+	.put  = scarlett2_mixer_ctl_put,
+	.private_value = SCARLETT2_MIXER_MAX_DB, /* max value */
+	.tlv = { .p = db_scale_scarlett2_mixer }
+};
+
+static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_ports *ports = private->info->ports;
+	int err, i, j;
+	int index;
+	char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+	int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT];
+	int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN];
+
+	for (i = 0, index = 0; i < num_outputs; i++) {
+		for (j = 0; j < num_inputs; j++, index++) {
+			snprintf(s, sizeof(s),
+				 "Mix %c Input %02d Playback Volume",
+				 'A' + i, j + 1);
+			err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl,
+						    index, 1, s, NULL);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	return 0;
+}
+
+/*** Mux Source Selection Controls ***/
+
+static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl,
+					   struct snd_ctl_elem_info *uinfo)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct scarlett2_mixer_data *private = elem->head.mixer->private_data;
+	const struct scarlett2_ports *ports = private->info->ports;
+	unsigned int item = uinfo->value.enumerated.item;
+	int items = private->num_mux_srcs;
+	int port_type;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = elem->channels;
+	uinfo->value.enumerated.items = items;
+
+	if (item >= items)
+		item = uinfo->value.enumerated.item = items - 1;
+
+	for (port_type = 0;
+	     port_type < SCARLETT2_PORT_TYPE_COUNT;
+	     port_type++) {
+		if (item < ports[port_type].num[SCARLETT2_PORT_IN]) {
+			sprintf(uinfo->value.enumerated.name,
+				ports[port_type].src_descr,
+				item + ports[port_type].src_num_offset);
+			return 0;
+		}
+		item -= ports[port_type].num[SCARLETT2_PORT_IN];
+	}
+
+	return -EINVAL;
+}
+
+static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl,
+					  struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct scarlett2_mixer_data *private = elem->head.mixer->private_data;
+
+	ucontrol->value.enumerated.item[0] = private->mux[elem->control];
+	return 0;
+}
+
+static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl,
+					  struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	struct usb_mixer_interface *mixer = elem->head.mixer;
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	int index = elem->control;
+	int oval, val, err = 0;
+
+	mutex_lock(&private->data_mutex);
+
+	oval = private->mux[index];
+	val = clamp(ucontrol->value.integer.value[0],
+		    0L, private->num_mux_srcs - 1L);
+
+	if (oval == val)
+		goto unlock;
+
+	private->mux[index] = val;
+	err = scarlett2_usb_set_mux(mixer);
+	if (err == 0)
+		err = 1;
+
+unlock:
+	mutex_unlock(&private->data_mutex);
+	return err;
+}
+
+static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "",
+	.info = scarlett2_mux_src_enum_ctl_info,
+	.get  = scarlett2_mux_src_enum_ctl_get,
+	.put  = scarlett2_mux_src_enum_ctl_put,
+};
+
+static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_ports *ports = private->info->ports;
+	int port_type, channel, i;
+
+	for (i = 0, port_type = 0;
+	     port_type < SCARLETT2_PORT_TYPE_COUNT;
+	     port_type++) {
+		for (channel = 0;
+		     channel < ports[port_type].num[SCARLETT2_PORT_OUT];
+		     channel++, i++) {
+			int err;
+			char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+			const char *const descr = ports[port_type].dst_descr;
+
+			snprintf(s, sizeof(s) - 5, descr, channel + 1);
+			strcat(s, " Enum");
+
+			err = scarlett2_add_new_ctl(mixer,
+						    &scarlett2_mux_src_enum_ctl,
+						    i, 1, s, NULL);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	return 0;
+}
+
+/*** Meter Controls ***/
+
+static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl,
+				    struct snd_ctl_elem_info *uinfo)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = elem->channels;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 4095;
+	uinfo->value.integer.step = 1;
+	return 0;
+}
+
+static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_info *elem = kctl->private_data;
+	u16 meter_levels[SCARLETT2_NUM_METERS];
+	int i, err;
+
+	err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels);
+	if (err < 0)
+		return err;
+
+	for (i = 0; i < elem->channels; i++)
+		ucontrol->value.integer.value[i] = meter_levels[i];
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new scarlett2_meter_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+	.name = "",
+	.info = scarlett2_meter_ctl_info,
+	.get  = scarlett2_meter_ctl_get
+};
+
+static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer)
+{
+	return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl,
+				     0, SCARLETT2_NUM_METERS,
+				     "Level Meter", NULL);
+}
+
+/*** Cleanup/Suspend Callbacks ***/
+
+static void scarlett2_private_free(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	cancel_delayed_work_sync(&private->work);
+	kfree(private);
+	mixer->private_data = NULL;
+}
+
+static void scarlett2_private_suspend(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+
+	if (cancel_delayed_work_sync(&private->work))
+		scarlett2_config_save(private->mixer);
+}
+
+/*** Initialisation ***/
+
+static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports)
+{
+	int port_type, count = 0;
+
+	for (port_type = 0;
+	     port_type < SCARLETT2_PORT_TYPE_COUNT;
+	     port_type++)
+		count += ports[port_type].num[SCARLETT2_PORT_IN];
+
+	return count;
+}
+
+/* Default routing connects PCM outputs and inputs to Analogue,
+ * S/PDIF, then ADAT
+ */
+static void scarlett2_init_routing(u8 *mux,
+				   const struct scarlett2_ports *ports)
+{
+	int i, input_num, input_count, port_type;
+	int output_num, output_count, port_type_connect_num;
+
+	static const int connect_order[] = {
+		SCARLETT2_PORT_TYPE_ANALOGUE,
+		SCARLETT2_PORT_TYPE_SPDIF,
+		SCARLETT2_PORT_TYPE_ADAT,
+		-1
+	};
+
+	/* Assign PCM inputs (routing outputs) */
+	output_num = scarlett2_get_port_start_num(ports,
+						  SCARLETT2_PORT_OUT,
+						  SCARLETT2_PORT_TYPE_PCM);
+	output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT];
+
+	for (port_type = connect_order[port_type_connect_num = 0];
+	     port_type >= 0;
+	     port_type = connect_order[++port_type_connect_num]) {
+		input_num = scarlett2_get_port_start_num(
+			ports, SCARLETT2_PORT_IN, port_type);
+		input_count = ports[port_type].num[SCARLETT2_PORT_IN];
+		for (i = 0;
+		     i < input_count && output_count;
+		     i++, output_count--)
+			mux[output_num++] = input_num++;
+	}
+
+	/* Assign PCM outputs (routing inputs) */
+	input_num = scarlett2_get_port_start_num(ports,
+						 SCARLETT2_PORT_IN,
+						 SCARLETT2_PORT_TYPE_PCM);
+	input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN];
+
+	for (port_type = connect_order[port_type_connect_num = 0];
+	     port_type >= 0;
+	     port_type = connect_order[++port_type_connect_num]) {
+		output_num = scarlett2_get_port_start_num(
+			ports, SCARLETT2_PORT_OUT, port_type);
+		output_count = ports[port_type].num[SCARLETT2_PORT_OUT];
+		for (i = 0;
+		     i < output_count && input_count;
+		     i++, input_count--)
+			mux[output_num++] = input_num++;
+	}
+}
+
+/* Initialise private data, routing, sequence number */
+static int scarlett2_init_private(struct usb_mixer_interface *mixer,
+				  const struct scarlett2_device_info *info)
+{
+	struct scarlett2_mixer_data *private =
+		kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL);
+
+	if (!private)
+		return -ENOMEM;
+
+	mutex_init(&private->usb_mutex);
+	mutex_init(&private->data_mutex);
+	INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work);
+	private->info = info;
+	private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports);
+	private->scarlett2_seq = 0;
+	private->mixer = mixer;
+	mixer->private_data = private;
+	mixer->private_free = scarlett2_private_free;
+	mixer->private_suspend = scarlett2_private_suspend;
+
+	/* Setup default routing */
+	scarlett2_init_routing(private->mux, info->ports);
+
+	/* Initialise the sequence number used for the proprietary commands */
+	return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, NULL, 0, NULL, 0);
+}
+
+/* Read line-in config and line-out volume settings on start */
+static int scarlett2_read_configs(struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_device_info *info = private->info;
+	const struct scarlett2_ports *ports = info->ports;
+	int num_line_out =
+		ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT];
+	u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX];
+	u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX];
+	struct scarlett2_usb_volume_status volume_status;
+	int err, i;
+
+	if (info->level_input_count) {
+		err = scarlett2_usb_get_config(
+			mixer,
+			SCARLETT2_CONFIG_LEVEL_SWITCH,
+			info->level_input_count,
+			level_switches);
+		if (err < 0)
+			return err;
+		for (i = 0; i < info->level_input_count; i++)
+			private->level_switch[i] = level_switches[i];
+	}
+
+	if (info->pad_input_count) {
+		err = scarlett2_usb_get_config(
+			mixer,
+			SCARLETT2_CONFIG_PAD_SWITCH,
+			info->pad_input_count,
+			pad_switches);
+		if (err < 0)
+			return err;
+		for (i = 0; i < info->pad_input_count; i++)
+			private->pad_switch[i] = pad_switches[i];
+	}
+
+	err = scarlett2_usb_get_volume_status(mixer, &volume_status);
+	if (err < 0)
+		return err;
+
+	private->master_vol = clamp(
+		volume_status.master_vol + SCARLETT2_VOLUME_BIAS,
+		0, SCARLETT2_VOLUME_BIAS);
+
+	for (i = 0; i < num_line_out; i++) {
+		int volume;
+
+		private->vol_sw_hw_switch[i] =
+			info->line_out_hw_vol
+				&& volume_status.sw_hw_switch[i];
+
+		volume = private->vol_sw_hw_switch[i]
+			   ? volume_status.master_vol
+			   : volume_status.sw_vol[i];
+		volume = clamp(volume + SCARLETT2_VOLUME_BIAS,
+			       0, SCARLETT2_VOLUME_BIAS);
+		private->vol[i] = volume;
+	}
+
+	for (i = 0; i < info->button_count; i++)
+		private->buttons[i] = !!volume_status.buttons[i];
+
+	return 0;
+}
+
+/* Notify on volume change */
+static void scarlett2_mixer_interrupt_vol_change(
+	struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	const struct scarlett2_ports *ports = private->info->ports;
+	int num_line_out =
+		ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT];
+	int i;
+
+	private->vol_updated = 1;
+
+	snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+		       &private->master_vol_ctl->id);
+
+	for (i = 0; i < num_line_out; i++) {
+		if (!private->vol_sw_hw_switch[i])
+			continue;
+		snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+			       &private->vol_ctls[i]->id);
+	}
+}
+
+/* Notify on button change */
+static void scarlett2_mixer_interrupt_button_change(
+	struct usb_mixer_interface *mixer)
+{
+	struct scarlett2_mixer_data *private = mixer->private_data;
+	int i;
+
+	private->vol_updated = 1;
+
+	for (i = 0; i < private->info->button_count; i++)
+		snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+			       &private->button_ctls[i]->id);
+}
+
+/* Interrupt callback */
+static void scarlett2_mixer_interrupt(struct urb *urb)
+{
+	struct usb_mixer_interface *mixer = urb->context;
+	int len = urb->actual_length;
+	int ustatus = urb->status;
+	u32 data;
+
+	if (ustatus != 0)
+		goto requeue;
+
+	if (len == 8) {
+		data = le32_to_cpu(*(u32 *)urb->transfer_buffer);
+		if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE)
+			scarlett2_mixer_interrupt_vol_change(mixer);
+		if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE)
+			scarlett2_mixer_interrupt_button_change(mixer);
+	} else {
+		usb_audio_err(mixer->chip,
+			      "scarlett mixer interrupt length %d\n", len);
+	}
+
+requeue:
+	if (ustatus != -ENOENT &&
+	    ustatus != -ECONNRESET &&
+	    ustatus != -ESHUTDOWN) {
+		urb->dev = mixer->chip->dev;
+		usb_submit_urb(urb, GFP_ATOMIC);
+	}
+}
+
+static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer)
+{
+	struct usb_device *dev = mixer->chip->dev;
+	unsigned int pipe = usb_rcvintpipe(dev,
+					   SCARLETT2_USB_INTERRUPT_ENDPOINT);
+	void *transfer_buffer;
+
+	if (mixer->urb) {
+		usb_audio_err(mixer->chip,
+			      "%s: mixer urb already in use!\n", __func__);
+		return 0;
+	}
+
+	if (snd_usb_pipe_sanity_check(dev, pipe))
+		return -EINVAL;
+
+	mixer->urb = usb_alloc_urb(0, GFP_KERNEL);
+	if (!mixer->urb)
+		return -ENOMEM;
+
+	transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL);
+	if (!transfer_buffer)
+		return -ENOMEM;
+
+	usb_fill_int_urb(mixer->urb, dev, pipe,
+			 transfer_buffer, SCARLETT2_USB_INTERRUPT_MAX_DATA,
+			 scarlett2_mixer_interrupt, mixer,
+			 SCARLETT2_USB_INTERRUPT_INTERVAL);
+
+	return usb_submit_urb(mixer->urb, GFP_KERNEL);
+}
+
+/* Entry point */
+int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer)
+{
+	const struct scarlett2_device_info *info;
+	int err;
+
+	/* only use UAC_VERSION_2 */
+	if (!mixer->protocol)
+		return 0;
+
+	switch (mixer->chip->usb_id) {
+	case USB_ID(0x1235, 0x8203):
+		info = &s6i6_gen2_info;
+		break;
+	case USB_ID(0x1235, 0x8204):
+		info = &s18i8_gen2_info;
+		break;
+	case USB_ID(0x1235, 0x8201):
+		info = &s18i20_gen2_info;
+		break;
+	default: /* device not (yet) supported */
+		return -EINVAL;
+	}
+
+	if (!(mixer->chip->setup & SCARLETT2_ENABLE)) {
+		usb_audio_err(mixer->chip,
+			"Focusrite Scarlett Gen 2 Mixer Driver disabled; "
+			"use options snd_usb_audio device_setup=1 "
+			"to enable and report any issues to g@b4.vu");
+		return 0;
+	}
+
+	/* Initialise private data, routing, sequence number */
+	err = scarlett2_init_private(mixer, info);
+	if (err < 0)
+		return err;
+
+	/* Read volume levels and controls from the interface */
+	err = scarlett2_read_configs(mixer);
+	if (err < 0)
+		return err;
+
+	/* Create the analogue output controls */
+	err = scarlett2_add_line_out_ctls(mixer);
+	if (err < 0)
+		return err;
+
+	/* Create the analogue input controls */
+	err = scarlett2_add_line_in_ctls(mixer);
+	if (err < 0)
+		return err;
+
+	/* Create the input, output, and mixer mux input selections */
+	err = scarlett2_add_mux_enums(mixer);
+	if (err < 0)
+		return err;
+
+	/* Create the matrix mixer controls */
+	err = scarlett2_add_mixer_ctls(mixer);
+	if (err < 0)
+		return err;
+
+	/* Create the level meter controls */
+	err = scarlett2_add_meter_ctl(mixer);
+	if (err < 0)
+		return err;
+
+	/* Set up the interrupt polling if there are hardware buttons */
+	if (info->button_count) {
+		err = scarlett2_mixer_status_create(mixer);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h
new file mode 100644
index 000000000000..52e1dad77afd
--- /dev/null
+++ b/sound/usb/mixer_scarlett_gen2.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __USB_MIXER_SCARLETT_GEN2_H
+#define __USB_MIXER_SCARLETT_GEN2_H
+
+int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer);
+
+#endif /* __USB_MIXER_SCARLETT_GEN2_H */
-- 
cgit v1.2.3-59-g8ed1b


From f474808acb3c4b30552d9c59b181244e0300d218 Mon Sep 17 00:00:00 2001
From: Johannes Berg <johannes@sipsolutions.net>
Date: Mon, 29 Jul 2019 09:14:22 +0200
Subject: ALSA: aoa: onyx: always initialize register read value

A lot of places in the driver use onyx_read_register() without
checking the return value, and it's been working OK for ~10 years
or so, so probably never fails ... Rather than trying to check the
return value everywhere, which would be relatively intrusive, at
least make sure we don't use an uninitialized value.

Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/aoa/codecs/onyx.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
index db917546965d..9827bee109c1 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -71,8 +71,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value)
 		return 0;
 	}
 	v = i2c_smbus_read_byte_data(onyx->i2c, reg);
-	if (v < 0)
+	if (v < 0) {
+		*value = 0;
 		return -1;
+	}
 	*value = (u8)v;
 	onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value;
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From 8e774e023565c682c67617da5141c7395c8a5fac Mon Sep 17 00:00:00 2001
From: "Gustavo A. R. Silva" <gustavo@embeddedor.com>
Date: Mon, 29 Jul 2019 15:54:54 -0500
Subject: sound: dmasound_atari: Mark expected switch fall-through

Mark switch cases where we are expecting to fall through.

This patch fixes the following warning (Building: m68k):

sound/oss/dmasound/dmasound_atari.c: warning: this statement may fall
through [-Wimplicit-fallthrough=]:  => 1449:24

Notice that, in this particular case, the code comment is
modified in accordance with what GCC is expecting to find.

Reported-by: Geert Uytterhoeven <geert@linux-m68k.org>
Tested-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Reviewed-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/dmasound/dmasound_atari.c | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 83653683fd68..823ccfa089b2 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -1432,25 +1432,25 @@ static int FalconMixerIoctl(u_int cmd, u_long arg)
 {
 	int data;
 	switch (cmd) {
-	    case SOUND_MIXER_READ_RECMASK:
+	case SOUND_MIXER_READ_RECMASK:
 		return IOCTL_OUT(arg, SOUND_MASK_MIC);
-	    case SOUND_MIXER_READ_DEVMASK:
+	case SOUND_MIXER_READ_DEVMASK:
 		return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC | SOUND_MASK_SPEAKER);
-	    case SOUND_MIXER_READ_STEREODEVS:
+	case SOUND_MIXER_READ_STEREODEVS:
 		return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC);
-	    case SOUND_MIXER_READ_VOLUME:
+	case SOUND_MIXER_READ_VOLUME:
 		return IOCTL_OUT(arg,
 			VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) |
 			VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8);
-	    case SOUND_MIXER_READ_CAPS:
+	case SOUND_MIXER_READ_CAPS:
 		return IOCTL_OUT(arg, SOUND_CAP_EXCL_INPUT);
-	    case SOUND_MIXER_WRITE_MIC:
+	case SOUND_MIXER_WRITE_MIC:
 		IOCTL_IN(arg, data);
 		tt_dmasnd.input_gain =
 			RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 |
 			RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff);
-		/* fall thru, return set value */
-	    case SOUND_MIXER_READ_MIC:
+		/* fall through - return set value */
+	case SOUND_MIXER_READ_MIC:
 		return IOCTL_OUT(arg,
 			RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) |
 			RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain & 0xf) << 8);
-- 
cgit v1.2.3-59-g8ed1b


From 83554cb94515467cfff10f70ec858d60bbbab8ca Mon Sep 17 00:00:00 2001
From: "Gustavo A. R. Silva" <gustavo@embeddedor.com>
Date: Mon, 29 Jul 2019 20:49:44 -0500
Subject: ALSA: sparc: Mark expected switch fall-throughs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Mark switch cases where we are expecting to fall through.

This patch fixes the following warnings (Building: sparc64):

sound/sparc/dbri.c: In function ‘reverse_bytes’:
sound/sparc/dbri.c:582:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
   b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16);
   ~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:583:2: note: here
  case 16:
  ^~~~
sound/sparc/dbri.c:584:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
   b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8);
   ~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:585:2: note: here
  case 8:
  ^~~~
sound/sparc/dbri.c:586:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
   b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4);
   ~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:587:2: note: here
  case 4:
  ^~~~
sound/sparc/dbri.c:588:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
   b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2);
   ~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:589:2: note: here
  case 2:
  ^~~~

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/sparc/dbri.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index 010113156239..6e065d44060e 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -580,12 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len)
 	switch (len) {
 	case 32:
 		b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16);
+		/* fall through */
 	case 16:
 		b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8);
+		/* fall through */
 	case 8:
 		b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4);
+		/* fall through */
 	case 4:
 		b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2);
+		/* fall through */
 	case 2:
 		b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1);
 	case 1:
-- 
cgit v1.2.3-59-g8ed1b


From 2c4956bc1e9062e5e3c5ea7612294f24e6d4fbdd Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Mon, 29 Jul 2019 10:01:39 +0200
Subject: ASoC: meson: g12a-tohdmitx: override codec2codec params

So far, forwarding the hw_params of the input to output relied on the
.hw_params() callback of the cpu side of the codec2codec link to be called
first. This is a bit weak.

Instead, override the stream params of the codec2codec to link to set it up
correctly.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190729080139.32068-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/g12a-tohdmitx.c | 34 ++++++++++++++++------------------
 1 file changed, 16 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c
index 707ccb192e4c..9943c807ec5d 100644
--- a/sound/soc/meson/g12a-tohdmitx.c
+++ b/sound/soc/meson/g12a-tohdmitx.c
@@ -28,7 +28,7 @@
 #define  CTRL0_SPDIF_CLK_SEL		BIT(0)
 
 struct g12a_tohdmitx_input {
-	struct snd_pcm_hw_params params;
+	struct snd_soc_pcm_stream params;
 	unsigned int fmt;
 };
 
@@ -225,26 +225,17 @@ static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream,
 {
 	struct g12a_tohdmitx_input *data = dai->playback_dma_data;
 
-	/* Save the stream params for the downstream link */
-	memcpy(&data->params, params, sizeof(*params));
+	data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params));
+	data->params.rate_min = params_rate(params);
+	data->params.rate_max = params_rate(params);
+	data->params.formats = 1 << params_format(params);
+	data->params.channels_min = params_channels(params);
+	data->params.channels_max = params_channels(params);
+	data->params.sig_bits = dai->driver->playback.sig_bits;
 
 	return 0;
 }
 
-static int g12a_tohdmitx_output_hw_params(struct snd_pcm_substream *substream,
-					  struct snd_pcm_hw_params *params,
-					  struct snd_soc_dai *dai)
-{
-	struct g12a_tohdmitx_input *in_data =
-		g12a_tohdmitx_get_input_data(dai->capture_widget);
-
-	if (!in_data)
-		return -ENODEV;
-
-	memcpy(params, &in_data->params, sizeof(*params));
-
-	return 0;
-}
 
 static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai,
 				       unsigned int fmt)
@@ -266,6 +257,14 @@ static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream,
 	if (!in_data)
 		return -ENODEV;
 
+	if (WARN_ON(!rtd->dai_link->params)) {
+		dev_warn(dai->dev, "codec2codec link expected\n");
+		return -EINVAL;
+	}
+
+	/* Replace link params with the input params */
+	rtd->dai_link->params = &in_data->params;
+
 	if (!in_data->fmt)
 		return 0;
 
@@ -278,7 +277,6 @@ static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = {
 };
 
 static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = {
-	.hw_params	= g12a_tohdmitx_output_hw_params,
 	.startup	= g12a_tohdmitx_output_startup,
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 21cc4ea85879c1b424fb01caa6e481c14286f7f0 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:05 +0800
Subject: ASoC: tegra20_das: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-2-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra20_das.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c
index 10f9c3b19c88..1070b2710d5e 100644
--- a/sound/soc/tegra/tegra20_das.c
+++ b/sound/soc/tegra/tegra20_das.c
@@ -120,7 +120,6 @@ static const struct regmap_config tegra20_das_regmap_config = {
 
 static int tegra20_das_probe(struct platform_device *pdev)
 {
-	struct resource *res;
 	void __iomem *regs;
 	int ret = 0;
 
@@ -134,8 +133,7 @@ static int tegra20_das_probe(struct platform_device *pdev)
 	}
 	das->dev = &pdev->dev;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(&pdev->dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs)) {
 		ret = PTR_ERR(regs);
 		goto err;
-- 
cgit v1.2.3-59-g8ed1b


From 7d22808641cdea50baeae636acf5d592a7aa1239 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:06 +0800
Subject: ASoC: tegra: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-3-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra30_i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index ac6983c6bd72..e6d548fa980b 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -368,7 +368,6 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev)
 	struct tegra30_i2s *i2s;
 	const struct of_device_id *match;
 	u32 cif_ids[2];
-	struct resource *mem;
 	void __iomem *regs;
 	int ret;
 
@@ -406,8 +405,7 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(&pdev->dev, mem);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs)) {
 		ret = PTR_ERR(regs);
 		goto err_clk_put;
-- 
cgit v1.2.3-59-g8ed1b


From f052172a5bb474fef2d0c3911655c9e107a3c024 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:07 +0800
Subject: ASoC: rockchip: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-4-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rk3328_codec.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c
index 24f8f86d58e9..287c962ba00d 100644
--- a/sound/soc/codecs/rk3328_codec.c
+++ b/sound/soc/codecs/rk3328_codec.c
@@ -432,7 +432,6 @@ static int rk3328_platform_probe(struct platform_device *pdev)
 {
 	struct device_node *rk3328_np = pdev->dev.of_node;
 	struct rk3328_codec_priv *rk3328;
-	struct resource *res;
 	struct regmap *grf;
 	void __iomem *base;
 	int ret = 0;
@@ -482,8 +481,7 @@ static int rk3328_platform_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 4b3540074361fc7b97f2da54365c55ceca6333d2 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:08 +0800
Subject: ASoC: ep93xx-i2s: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-5-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/cirrus/ep93xx-i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index 0b4355e95f84..7d9cf67129d4 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -430,15 +430,13 @@ static const struct snd_soc_component_driver ep93xx_i2s_component = {
 static int ep93xx_i2s_probe(struct platform_device *pdev)
 {
 	struct ep93xx_i2s_info *info;
-	struct resource *res;
 	int err;
 
 	info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
 	if (!info)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	info->regs = devm_ioremap_resource(&pdev->dev, res);
+	info->regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(info->regs))
 		return PTR_ERR(info->regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From fceef72b68d63593d8543a8d608ccc8427e13d56 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:09 +0800
Subject: ASoC: mt8173: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-6-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
index 0382896c162e..90bd2c92cae7 100644
--- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
+++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
@@ -1056,7 +1056,6 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev)
 	int irq_id;
 	struct mtk_base_afe *afe;
 	struct mt8173_afe_private *afe_priv;
-	struct resource *res;
 
 	ret = dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(33));
 	if (ret)
@@ -1086,8 +1085,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	afe->base_addr = devm_ioremap_resource(&pdev->dev, res);
+	afe->base_addr = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(afe->base_addr))
 		return PTR_ERR(afe->base_addr);
 
-- 
cgit v1.2.3-59-g8ed1b


From 61e799e3658952397da80f6c2e3581e7b65209a8 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:10 +0800
Subject: ASoC: mt6797: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-7-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
index 08a6532da322..e52c032d53aa 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
@@ -749,7 +749,6 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
 {
 	struct mtk_base_afe *afe;
 	struct mt6797_afe_private *afe_priv;
-	struct resource *res;
 	struct device *dev;
 	int i, irq_id, ret;
 
@@ -774,9 +773,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev)
 	}
 
 	/* regmap init */
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-
-	afe->base_addr = devm_ioremap_resource(&pdev->dev, res);
+	afe->base_addr = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(afe->base_addr))
 		return PTR_ERR(afe->base_addr);
 
-- 
cgit v1.2.3-59-g8ed1b


From d003e3081c1d069b94d1b5f4c3bc8234d4603282 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:11 +0800
Subject: ASoC: imx-audmux: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-8-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/imx-audmux.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index b2351cd33b0f..7595f24a006e 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -298,12 +298,10 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev,
 
 static int imx_audmux_probe(struct platform_device *pdev)
 {
-	struct resource *res;
 	const struct of_device_id *of_id =
 			of_match_device(imx_audmux_dt_ids, &pdev->dev);
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	audmux_base = devm_ioremap_resource(&pdev->dev, res);
+	audmux_base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(audmux_base))
 		return PTR_ERR(audmux_base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 959bb6b54d708652fe06097feecc3acd4d6b8c89 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:12 +0800
Subject: ASoC: fsl_audmix: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-9-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_audmix.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
index 3897a54a11fe..c7e4e9757dce 100644
--- a/sound/soc/fsl/fsl_audmix.c
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -458,7 +458,6 @@ static int fsl_audmix_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
 	struct fsl_audmix *priv;
-	struct resource *res;
 	const char *mdrv;
 	const struct of_device_id *of_id;
 	void __iomem *regs;
@@ -475,8 +474,7 @@ static int fsl_audmix_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	/* Get the addresses */
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs))
 		return PTR_ERR(regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From d400b1b30cbff1c0962bbb3f895e839ccd120cb7 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:13 +0800
Subject: ASoC: bcm2835-i2s: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-10-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/bcm/bcm2835-i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c
index 5ef80f3d446a..e6a12e271b07 100644
--- a/sound/soc/bcm/bcm2835-i2s.c
+++ b/sound/soc/bcm/bcm2835-i2s.c
@@ -828,7 +828,6 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
 {
 	struct bcm2835_i2s_dev *dev;
 	int ret;
-	struct resource *mem;
 	void __iomem *base;
 	const __be32 *addr;
 	dma_addr_t dma_base;
@@ -848,8 +847,7 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
 	}
 
 	/* Request ioarea */
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, mem);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
-- 
cgit v1.2.3-59-g8ed1b


From c9dbe688983b1d77a9a46cf6b7ac88348a1ca395 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:14 +0800
Subject: ASoC: sirf: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-11-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sirf/sirf-usp.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c
index 8bab119c753a..2af0c6f14ee6 100644
--- a/sound/soc/sirf/sirf-usp.c
+++ b/sound/soc/sirf/sirf-usp.c
@@ -359,7 +359,6 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev)
 	int ret;
 	struct sirf_usp *usp;
 	void __iomem *base;
-	struct resource *mem_res;
 
 	usp = devm_kzalloc(&pdev->dev, sizeof(struct sirf_usp),
 			GFP_KERNEL);
@@ -368,8 +367,7 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(pdev, usp);
 
-	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, mem_res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 	usp->regmap = devm_regmap_init_mmio(&pdev->dev, base,
-- 
cgit v1.2.3-59-g8ed1b


From 1327bfe2887c0e29bcdc0c5c3e445da107ccb4e2 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:15 +0800
Subject: ASoC: mxs-saif: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-12-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mxs/mxs-saif.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 269b6d6df250..a2c79426513b 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -732,7 +732,6 @@ static int mxs_saif_mclk_init(struct platform_device *pdev)
 static int mxs_saif_probe(struct platform_device *pdev)
 {
 	struct device_node *np = pdev->dev.of_node;
-	struct resource *iores;
 	struct mxs_saif *saif;
 	int irq, ret = 0;
 	struct device_node *master;
@@ -786,9 +785,7 @@ static int mxs_saif_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-
-	saif->base = devm_ioremap_resource(&pdev->dev, iores);
+	saif->base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(saif->base))
 		return PTR_ERR(saif->base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 0fb46f541bb3e6e5bbb155c08b9e75f92c21acb9 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:16 +0800
Subject: ASoC: spear: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-13-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/spear/spdif_in.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 78a6a360b4a6..4b68d6ee75da 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -202,12 +202,11 @@ static int spdif_in_probe(struct platform_device *pdev)
 {
 	struct spdif_in_dev *host;
 	struct spear_spdif_platform_data *pdata;
-	struct resource *res, *res_fifo;
+	struct resource *res_fifo;
 	void __iomem *io_base;
 	int ret;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	io_base = devm_ioremap_resource(&pdev->dev, res);
+	io_base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(io_base))
 		return PTR_ERR(io_base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 89dd38bf130e661d92c6ab7e3b003907f1d3f367 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:17 +0800
Subject: ASoC: kirkwood-i2s: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-14-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/kirkwood/kirkwood-i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3446a113f482..5076ec4cc7a6 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -523,7 +523,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data;
 	struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai;
 	struct kirkwood_dma_data *priv;
-	struct resource *mem;
 	struct device_node *np = pdev->dev.of_node;
 	int err;
 
@@ -533,8 +532,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 
 	dev_set_drvdata(&pdev->dev, priv);
 
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	priv->io = devm_ioremap_resource(&pdev->dev, mem);
+	priv->io = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(priv->io))
 		return PTR_ERR(priv->io);
 
-- 
cgit v1.2.3-59-g8ed1b


From ebdd7be5415c7795c77609ad908222038e441835 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:18 +0800
Subject: ASoC: xtfpga-i2s: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-15-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/xtensa/xtfpga-i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c
index 9ce2c75186b9..9da395d14a8d 100644
--- a/sound/soc/xtensa/xtfpga-i2s.c
+++ b/sound/soc/xtensa/xtfpga-i2s.c
@@ -531,7 +531,6 @@ static int xtfpga_i2s_runtime_resume(struct device *dev)
 static int xtfpga_i2s_probe(struct platform_device *pdev)
 {
 	struct xtfpga_i2s *i2s;
-	struct resource *mem;
 	int err, irq;
 
 	i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
@@ -543,8 +542,7 @@ static int xtfpga_i2s_probe(struct platform_device *pdev)
 	i2s->dev = &pdev->dev;
 	dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s);
 
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	i2s->regs = devm_ioremap_resource(&pdev->dev, mem);
+	i2s->regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(i2s->regs)) {
 		err = PTR_ERR(i2s->regs);
 		goto err;
-- 
cgit v1.2.3-59-g8ed1b


From 22581e7c51f50fbbadba70499bb5e2d09f830cbb Mon Sep 17 00:00:00 2001
From: "Gustavo A. R. Silva" <gustavo@embeddedor.com>
Date: Mon, 29 Jul 2019 17:15:34 -0500
Subject: ASoC: ti: Mark expected switch fall-throughs
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Mark switch cases where we are expecting to fall through.

This patch fixes the following warning (Building: arm):

sound/soc/ti/n810.c: In function ‘n810_ext_control’:
sound/soc/ti/n810.c:48:10: warning: this statement may fall through [-Wimplicit-fallthrough=]
   line1l = 1;
   ~~~~~~~^~~
sound/soc/ti/n810.c:49:2: note: here
  case N810_JACK_HP:
  ^~~~

sound/soc/ti/rx51.c: In function ‘rx51_ext_control’:
sound/soc/ti/rx51.c:57:6: warning: this statement may fall through [-Wimplicit-fallthrough=]
   hs = 1;
   ~~~^~~
sound/soc/ti/rx51.c:58:2: note: here
  case RX51_JACK_HP:
  ^~~~

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Link: https://lore.kernel.org/r/20190729221534.GA18696@embeddedor
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/n810.c | 1 +
 sound/soc/ti/rx51.c | 1 +
 2 files changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 2c3f2a4c1700..3ad2b6daf31e 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -46,6 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
 	switch (n810_jack_func) {
 	case N810_JACK_HS:
 		line1l = 1;
+		/* fall through */
 	case N810_JACK_HP:
 		hp = 1;
 		break;
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index bc6046534fa5..ccd0e8a07dd1 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -55,6 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
 		break;
 	case RX51_JACK_HS:
 		hs = 1;
+		/* fall through */
 	case RX51_JACK_HP:
 		hp = 1;
 		break;
-- 
cgit v1.2.3-59-g8ed1b


From 0c03e37af47efcb8600f95f399783c082fcf2f93 Mon Sep 17 00:00:00 2001
From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Date: Wed, 24 Jul 2019 08:00:23 +0200
Subject: ASoC: cs4271: Fix a typo in the CS4171_NR_RATIOS

This should be CS4271_NR_RATIOS.
Fix it and use it.

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20190724060023.31302-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs4271.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 1d03a1348162..04b86a51e055 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -334,7 +334,7 @@ static struct cs4271_clk_cfg cs4271_clk_tab[] = {
 	{0, CS4271_MODE1_MODE_4X, 256,  CS4271_MODE1_DIV_2},
 };
 
-#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
+#define CS4271_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
 
 static int cs4271_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_pcm_hw_params *params,
@@ -383,13 +383,13 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
 		val = CS4271_MODE1_MODE_4X;
 
 	ratio = cs4271->mclk / cs4271->rate;
-	for (i = 0; i < CS4171_NR_RATIOS; i++)
+	for (i = 0; i < CS4271_NR_RATIOS; i++)
 		if ((cs4271_clk_tab[i].master == cs4271->master) &&
 		    (cs4271_clk_tab[i].speed_mode == val) &&
 		    (cs4271_clk_tab[i].ratio == ratio))
 			break;
 
-	if (i == CS4171_NR_RATIOS) {
+	if (i == CS4271_NR_RATIOS) {
 		dev_err(component->dev, "Invalid sample rate\n");
 		return -EINVAL;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From 2e73d98f2ecf61008926bb8a425c3281a89959c0 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:31 +0800
Subject: ASoC: meson: axg-pdm: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-28-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-pdm.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c
index 9d5684493ffc..bfd37d49a73e 100644
--- a/sound/soc/meson/axg-pdm.c
+++ b/sound/soc/meson/axg-pdm.c
@@ -585,7 +585,6 @@ static int axg_pdm_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
 	struct axg_pdm *priv;
-	struct resource *res;
 	void __iomem *regs;
 	int ret;
 
@@ -600,8 +599,7 @@ static int axg_pdm_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs))
 		return PTR_ERR(regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From d61feec0f60c250c786a9e0069ce909979491c7b Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:32 +0800
Subject: ASoC: meson: axg-spdifin: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-29-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-spdifin.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c
index 01b2035fa841..d0d09f945b48 100644
--- a/sound/soc/meson/axg-spdifin.c
+++ b/sound/soc/meson/axg-spdifin.c
@@ -453,7 +453,6 @@ static int axg_spdifin_probe(struct platform_device *pdev)
 	struct device *dev = &pdev->dev;
 	struct axg_spdifin *priv;
 	struct snd_soc_dai_driver *dai_drv;
-	struct resource *res;
 	void __iomem *regs;
 	int ret;
 
@@ -468,8 +467,7 @@ static int axg_spdifin_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs))
 		return PTR_ERR(regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From c43fd289dd58951f3cfddb3bb66a6ed7747c5986 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:33 +0800
Subject: ASoC: meson: axg-spdifout: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-30-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-spdifout.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c
index 9dea528053ad..7ce6aa97ddf7 100644
--- a/sound/soc/meson/axg-spdifout.c
+++ b/sound/soc/meson/axg-spdifout.c
@@ -401,7 +401,6 @@ static int axg_spdifout_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
 	struct axg_spdifout *priv;
-	struct resource *res;
 	void __iomem *regs;
 	int ret;
 
@@ -410,8 +409,7 @@ static int axg_spdifout_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	platform_set_drvdata(pdev, priv);
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs))
 		return PTR_ERR(regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From 9b2089313156ad16bd257217cc67e10597a82923 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:34 +0800
Subject: ASoC: meson: axg-fifo: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-31-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-fifo.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 01c1c7db2510..80a3dde35b5c 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -314,7 +314,6 @@ int axg_fifo_probe(struct platform_device *pdev)
 	struct device *dev = &pdev->dev;
 	const struct axg_fifo_match_data *data;
 	struct axg_fifo *fifo;
-	struct resource *res;
 	void __iomem *regs;
 
 	data = of_device_get_match_data(dev);
@@ -328,8 +327,7 @@ int axg_fifo_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	platform_set_drvdata(pdev, fifo);
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs))
 		return PTR_ERR(regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From cf80aa2c1359f5d014981e251049bcc21a2217bc Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:35 +0800
Subject: ASoC: xlnx: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-32-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/xilinx/xlnx_i2s.c   | 4 +---
 sound/soc/xilinx/xlnx_spdif.c | 3 +--
 2 files changed, 2 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c
index 8b353166ad44..cc641e582c82 100644
--- a/sound/soc/xilinx/xlnx_i2s.c
+++ b/sound/soc/xilinx/xlnx_i2s.c
@@ -95,7 +95,6 @@ MODULE_DEVICE_TABLE(of, xlnx_i2s_of_match);
 
 static int xlnx_i2s_probe(struct platform_device *pdev)
 {
-	struct resource *res;
 	void __iomem *base;
 	struct snd_soc_dai_driver *dai_drv;
 	int ret;
@@ -107,8 +106,7 @@ static int xlnx_i2s_probe(struct platform_device *pdev)
 	if (!dai_drv)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
diff --git a/sound/soc/xilinx/xlnx_spdif.c b/sound/soc/xilinx/xlnx_spdif.c
index 3b9000fd8c49..e2ca087adee6 100644
--- a/sound/soc/xilinx/xlnx_spdif.c
+++ b/sound/soc/xilinx/xlnx_spdif.c
@@ -260,8 +260,7 @@ static int xlnx_spdif_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	ctx->base = devm_ioremap_resource(dev, res);
+	ctx->base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(ctx->base)) {
 		ret = PTR_ERR(ctx->base);
 		goto clk_err;
-- 
cgit v1.2.3-59-g8ed1b


From e12fd61fd7c1bbbb758ede9fdf58ac2bd5de6b33 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:36 +0800
Subject: ASoC: sun8i-codec-analog: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-33-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun8i-codec-analog.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c
index e92aeedd6feb..be872eefa61e 100644
--- a/sound/soc/sunxi/sun8i-codec-analog.c
+++ b/sound/soc/sunxi/sun8i-codec-analog.c
@@ -819,12 +819,10 @@ MODULE_DEVICE_TABLE(of, sun8i_codec_analog_of_match);
 
 static int sun8i_codec_analog_probe(struct platform_device *pdev)
 {
-	struct resource *res;
 	struct regmap *regmap;
 	void __iomem *base;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base)) {
 		dev_err(&pdev->dev, "Failed to map the registers\n");
 		return PTR_ERR(base);
-- 
cgit v1.2.3-59-g8ed1b


From 54dd39e4aca8edb231e13349ac3a3a44ef28d3d4 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:37 +0800
Subject: ASoC: sunxi: sun50i-codec-analog: use
 devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-34-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun50i-codec-analog.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
index 6d1de565350e..f5b7069bcca2 100644
--- a/sound/soc/sunxi/sun50i-codec-analog.c
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -459,12 +459,10 @@ MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match);
 
 static int sun50i_codec_analog_probe(struct platform_device *pdev)
 {
-	struct resource *res;
 	struct regmap *regmap;
 	void __iomem *base;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base)) {
 		dev_err(&pdev->dev, "Failed to map the registers\n");
 		return PTR_ERR(base);
-- 
cgit v1.2.3-59-g8ed1b


From 790b36578e23264ff05be2ba8466ebcf7d3db38a Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:38 +0800
Subject: ASoC: sun8i-codec: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-35-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun8i-codec.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 0e0e8ebaa571..55798bc8eae2 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -533,7 +533,6 @@ static const struct regmap_config sun8i_codec_regmap_config = {
 
 static int sun8i_codec_probe(struct platform_device *pdev)
 {
-	struct resource *res_base;
 	struct sun8i_codec *scodec;
 	void __iomem *base;
 	int ret;
@@ -556,8 +555,7 @@ static int sun8i_codec_probe(struct platform_device *pdev)
 		return PTR_ERR(scodec->clk_bus);
 	}
 
-	res_base = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, res_base);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base)) {
 		dev_err(&pdev->dev, "Failed to map the registers\n");
 		return PTR_ERR(base);
-- 
cgit v1.2.3-59-g8ed1b


From 1c63223cda84fb10643938e4e4eb02110f8ae9a2 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:19 +0800
Subject: ASoC: stm32: sai: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-16-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/stm/stm32_sai.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index 63f68e663676..1ac5103cea78 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -152,7 +152,6 @@ static int stm32_sai_probe(struct platform_device *pdev)
 {
 	struct stm32_sai_data *sai;
 	struct reset_control *rst;
-	struct resource *res;
 	const struct of_device_id *of_id;
 	u32 val;
 	int ret;
@@ -161,8 +160,7 @@ static int stm32_sai_probe(struct platform_device *pdev)
 	if (!sai)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	sai->base = devm_ioremap_resource(&pdev->dev, res);
+	sai->base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(sai->base))
 		return PTR_ERR(sai->base);
 
-- 
cgit v1.2.3-59-g8ed1b


From f62d6426718837663484bbf0ed94e56573cbc365 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:20 +0800
Subject: ASoC: codecs: msm8916-wcd: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-17-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/msm8916-wcd-digital.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 1db7e43ec203..9fa5d44fdc79 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -1143,7 +1143,6 @@ static int msm8916_wcd_digital_probe(struct platform_device *pdev)
 	struct msm8916_wcd_digital_priv *priv;
 	struct device *dev = &pdev->dev;
 	void __iomem *base;
-	struct resource *mem_res;
 	struct regmap *digital_map;
 	int ret;
 
@@ -1151,8 +1150,7 @@ static int msm8916_wcd_digital_probe(struct platform_device *pdev)
 	if (!priv)
 		return -ENOMEM;
 
-	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, mem_res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 94db63761576a98279e19f51173dec85f803bad4 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:21 +0800
Subject: ASoC: codecs: jz4725b: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-18-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/jz4725b.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c
index 766354c73076..2567a5d15b55 100644
--- a/sound/soc/codecs/jz4725b.c
+++ b/sound/soc/codecs/jz4725b.c
@@ -545,15 +545,13 @@ static int jz4725b_codec_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
 	struct jz_icdc *icdc;
-	struct resource *mem;
 	int ret;
 
 	icdc = devm_kzalloc(dev, sizeof(*icdc), GFP_KERNEL);
 	if (!icdc)
 		return -ENOMEM;
 
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	icdc->base = devm_ioremap_resource(dev, mem);
+	icdc->base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(icdc->base))
 		return PTR_ERR(icdc->base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 72d09322e709c5c2d7112d6a19ec9f3846d3609d Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:22 +0800
Subject: ASoC: mmp-sspa: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-19-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/pxa/mmp-sspa.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index 72f4364b2d20..e3e5425b5c62 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -399,7 +399,6 @@ static const struct snd_soc_component_driver mmp_sspa_component = {
 static int asoc_mmp_sspa_probe(struct platform_device *pdev)
 {
 	struct sspa_priv *priv;
-	struct resource *res;
 
 	priv = devm_kzalloc(&pdev->dev,
 				sizeof(struct sspa_priv), GFP_KERNEL);
@@ -417,8 +416,7 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev)
 	if (priv->dma_params == NULL)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res);
+	priv->sspa->mmio_base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(priv->sspa->mmio_base))
 		return PTR_ERR(priv->sspa->mmio_base);
 
-- 
cgit v1.2.3-59-g8ed1b


From a8dc106410a2c3c87917447fb8c132b66062443e Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:23 +0800
Subject: ASoC: jz4740: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-20-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/jz4740.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 974e17fa1911..460aa1fd1efe 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -318,7 +318,6 @@ static int jz4740_codec_probe(struct platform_device *pdev)
 {
 	int ret;
 	struct jz4740_codec *jz4740_codec;
-	struct resource *mem;
 	void __iomem *base;
 
 	jz4740_codec = devm_kzalloc(&pdev->dev, sizeof(*jz4740_codec),
@@ -326,8 +325,7 @@ static int jz4740_codec_probe(struct platform_device *pdev)
 	if (!jz4740_codec)
 		return -ENOMEM;
 
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, mem);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 2d1ffc7d0f58f6a39a34854e952f5020fc655af0 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:24 +0800
Subject: ASoC: inno_rk3036: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-21-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/inno_rk3036.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c
index 7feedbb7bbed..14d8fe1c28a4 100644
--- a/sound/soc/codecs/inno_rk3036.c
+++ b/sound/soc/codecs/inno_rk3036.c
@@ -405,7 +405,6 @@ static int rk3036_codec_platform_probe(struct platform_device *pdev)
 {
 	struct rk3036_codec_priv *priv;
 	struct device_node *of_node = pdev->dev.of_node;
-	struct resource *res;
 	void __iomem *base;
 	struct regmap *grf;
 	int ret;
@@ -414,8 +413,7 @@ static int rk3036_codec_platform_probe(struct platform_device *pdev)
 	if (!priv)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
-- 
cgit v1.2.3-59-g8ed1b


From 4e5bc35988af2483138feb3c5c3bf5fb0cae4228 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:25 +0800
Subject: ASoC: uniphier: evea: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-22-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/uniphier/evea.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c
index f9c10165fbc1..d27e9ca07856 100644
--- a/sound/soc/uniphier/evea.c
+++ b/sound/soc/uniphier/evea.c
@@ -451,7 +451,6 @@ static const struct regmap_config evea_regmap_config = {
 static int evea_probe(struct platform_device *pdev)
 {
 	struct evea_priv *evea;
-	struct resource *res;
 	void __iomem *preg;
 	int ret;
 
@@ -475,8 +474,7 @@ static int evea_probe(struct platform_device *pdev)
 	if (IS_ERR(evea->rst_exiv))
 		return PTR_ERR(evea->rst_exiv);
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	preg = devm_ioremap_resource(&pdev->dev, res);
+	preg = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(preg))
 		return PTR_ERR(preg);
 
-- 
cgit v1.2.3-59-g8ed1b


From b885c9fa363fa4604d78ea00adfed64db656fb78 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:26 +0800
Subject: ASoC: uniphier: aio-dma: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-23-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/uniphier/aio-dma.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c
index fa001d3c1a88..862346d66774 100644
--- a/sound/soc/uniphier/aio-dma.c
+++ b/sound/soc/uniphier/aio-dma.c
@@ -276,12 +276,10 @@ int uniphier_aiodma_soc_register_platform(struct platform_device *pdev)
 {
 	struct uniphier_aio_chip *chip = platform_get_drvdata(pdev);
 	struct device *dev = &pdev->dev;
-	struct resource *res;
 	void __iomem *preg;
 	int irq, ret;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	preg = devm_ioremap_resource(dev, res);
+	preg = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(preg))
 		return PTR_ERR(preg);
 
-- 
cgit v1.2.3-59-g8ed1b


From b1abe8780041d568ae07013a3592519fca855a1c Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:27 +0800
Subject: ASoC: psc-ac97: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-24-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/au1x/psc-ac97.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 21e5f6aed7f3..08bc04e2da2a 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -363,7 +363,7 @@ static const struct snd_soc_component_driver au1xpsc_ac97_component = {
 static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 {
 	int ret;
-	struct resource *iores, *dmares;
+	struct resource *dmares;
 	unsigned long sel;
 	struct au1xpsc_audio_data *wd;
 
@@ -374,8 +374,7 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 
 	mutex_init(&wd->lock);
 
-	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	wd->mmio = devm_ioremap_resource(&pdev->dev, iores);
+	wd->mmio = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(wd->mmio))
 		return PTR_ERR(wd->mmio);
 
-- 
cgit v1.2.3-59-g8ed1b


From 12a63c0fa03691328b948690601dc7dde8fc527b Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:28 +0800
Subject: ASoC: au1x: psc-i2s: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-25-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/au1x/psc-i2s.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 076303f96b8c..767ce950d0da 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -291,7 +291,7 @@ static const struct snd_soc_component_driver au1xpsc_i2s_component = {
 
 static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 {
-	struct resource *iores, *dmares;
+	struct resource *dmares;
 	unsigned long sel;
 	struct au1xpsc_audio_data *wd;
 
@@ -300,8 +300,7 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 	if (!wd)
 		return -ENOMEM;
 
-	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	wd->mmio = devm_ioremap_resource(&pdev->dev, iores);
+	wd->mmio = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(wd->mmio))
 		return PTR_ERR(wd->mmio);
 
-- 
cgit v1.2.3-59-g8ed1b


From a95c901248642d62dc7462a2d2190c8ac183c84f Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:29 +0800
Subject: ASoC: meson: g12a-tohdmitx: use devm_platform_ioremap_resource() to
 simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-26-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/g12a-tohdmitx.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c
index 9943c807ec5d..9cfbd343a00c 100644
--- a/sound/soc/meson/g12a-tohdmitx.c
+++ b/sound/soc/meson/g12a-tohdmitx.c
@@ -376,12 +376,10 @@ MODULE_DEVICE_TABLE(of, g12a_tohdmitx_of_match);
 static int g12a_tohdmitx_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
-	struct resource *res;
 	void __iomem *regs;
 	struct regmap *map;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs))
 		return PTR_ERR(regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From 10392fcad7dfc8ea38959b18327ff18b81b1c161 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Sat, 27 Jul 2019 23:07:30 +0800
Subject: ASoC: meson: axg-tdm-formatter: use devm_platform_ioremap_resource()
 to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190727150738.54764-27-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-tdm-formatter.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 2e498201139f..21c735afab35 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -253,7 +253,6 @@ int axg_tdm_formatter_probe(struct platform_device *pdev)
 	struct device *dev = &pdev->dev;
 	const struct axg_tdm_formatter_driver *drv;
 	struct axg_tdm_formatter *formatter;
-	struct resource *res;
 	void __iomem *regs;
 	int ret;
 
@@ -269,8 +268,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, formatter);
 	formatter->drv = drv;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	regs = devm_ioremap_resource(dev, res);
+	regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(regs))
 		return PTR_ERR(regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From caa918ef14065b812737f3ee4ac349dcfff196e6 Mon Sep 17 00:00:00 2001
From: Andra Danciu <andradanciu1997@gmail.com>
Date: Wed, 31 Jul 2019 14:19:30 +0300
Subject: ASoC: codecs: Add uda1334 codec driver

The UDA1334BTS supports the I2S-bus data format with word lengths of up
to 24 bits serial data format and has basic features such as de-emphasis
(at 44.1 kHz sampling rate) and mute.

Datasheet can be found at:
https://www.nxp.com/docs/en/data-sheet/UDA1334BTS.pdf

Cc: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Link: https://lore.kernel.org/r/20190731111930.20230-3-andradanciu1997@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/Kconfig   |   9 ++
 sound/soc/codecs/Makefile  |   2 +
 sound/soc/codecs/uda1334.c | 295 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 306 insertions(+)
 create mode 100644 sound/soc/codecs/uda1334.c

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index da4c1ae89742..89238343e34d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -199,6 +199,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TS3A227E if I2C
 	select SND_SOC_TWL4030 if TWL4030_CORE
 	select SND_SOC_TWL6040 if TWL6040_CORE
+	select SND_SOC_UDA1334 if GPIOLIB
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
 	select SND_SOC_WCD9335 if SLIMBUS
@@ -1207,6 +1208,14 @@ config SND_SOC_TWL4030
 config SND_SOC_TWL6040
 	tristate
 
+config SND_SOC_UDA1334
+	tristate "NXP UDA1334 DAC"
+	depends on GPIOLIB
+	help
+	  The UDA1334 is an NXP audio codec, supports the I2S-bus data format
+	  and has basic features such as de-emphasis (at 44.1 kHz sampling
+	  rate) and mute.
+
 config SND_SOC_UDA134X
        tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9230016b0f9f..c498373dcc5f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -212,6 +212,7 @@ snd-soc-tscs454-objs := tscs454.o
 snd-soc-ts3a227e-objs := ts3a227e.o
 snd-soc-twl4030-objs := twl4030.o
 snd-soc-twl6040-objs := twl6040.o
+snd-soc-uda1334-objs := uda1334.o
 snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
 snd-soc-wcd9335-objs := wcd-clsh-v2.o wcd9335.o
@@ -494,6 +495,7 @@ obj-$(CONFIG_SND_SOC_TSCS454)	+= snd-soc-tscs454.o
 obj-$(CONFIG_SND_SOC_TS3A227E)	+= snd-soc-ts3a227e.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
+obj-$(CONFIG_SND_SOC_UDA1334)	+= snd-soc-uda1334.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
 obj-$(CONFIG_SND_SOC_WCD9335)	+= snd-soc-wcd9335.o
diff --git a/sound/soc/codecs/uda1334.c b/sound/soc/codecs/uda1334.c
new file mode 100644
index 000000000000..21ab8c5487ba
--- /dev/null
+++ b/sound/soc/codecs/uda1334.c
@@ -0,0 +1,295 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// uda1334.c  --  UDA1334 ALSA SoC Audio driver
+//
+// Based on WM8523 ALSA SoC Audio driver written by Mark Brown
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/gpio/consumer.h>
+#include <linux/of_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#define UDA1334_NUM_RATES 6
+
+/* codec private data */
+struct uda1334_priv {
+	struct gpio_desc *mute;
+	struct gpio_desc *deemph;
+	unsigned int sysclk;
+	unsigned int rate_constraint_list[UDA1334_NUM_RATES];
+	struct snd_pcm_hw_constraint_list rate_constraint;
+};
+
+static const struct snd_soc_dapm_widget uda1334_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("LINEVOUTL"),
+SND_SOC_DAPM_OUTPUT("LINEVOUTR"),
+};
+
+static const struct snd_soc_dapm_route uda1334_dapm_routes[] = {
+	{ "LINEVOUTL", NULL, "DAC" },
+	{ "LINEVOUTR", NULL, "DAC" },
+};
+
+static int uda1334_put_deemph(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct uda1334_priv *uda1334 = snd_soc_component_get_drvdata(component);
+	int deemph = ucontrol->value.integer.value[0];
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	gpiod_set_value_cansleep(uda1334->deemph, deemph);
+
+	return 0;
+};
+
+static int uda1334_get_deemph(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct uda1334_priv *uda1334 = snd_soc_component_get_drvdata(component);
+	int ret;
+
+	ret = gpiod_get_value_cansleep(uda1334->deemph);
+	if (ret < 0)
+		return -EINVAL;
+
+	ucontrol->value.integer.value[0] = ret;
+
+	return 0;
+};
+
+static const struct snd_kcontrol_new uda1334_snd_controls[] = {
+	SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
+			    uda1334_get_deemph, uda1334_put_deemph),
+};
+
+static const struct {
+	int value;
+	int ratio;
+} lrclk_ratios[UDA1334_NUM_RATES] = {
+	{ 1, 128 },
+	{ 2, 192 },
+	{ 3, 256 },
+	{ 4, 384 },
+	{ 5, 512 },
+	{ 6, 768 },
+};
+
+static int uda1334_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct uda1334_priv *uda1334 = snd_soc_component_get_drvdata(component);
+
+	/*
+	 * The set of sample rates that can be supported depends on the
+	 * MCLK supplied to the CODEC - enforce this.
+	 */
+	if (!uda1334->sysclk) {
+		dev_err(component->dev,
+			"No MCLK configured, call set_sysclk() on init\n");
+		return -EINVAL;
+	}
+
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_RATE,
+				   &uda1334->rate_constraint);
+
+	gpiod_set_value_cansleep(uda1334->mute, 1);
+
+	return 0;
+}
+
+static void uda1334_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct uda1334_priv *uda1334 = snd_soc_component_get_drvdata(component);
+
+	gpiod_set_value_cansleep(uda1334->mute, 0);
+}
+
+static int uda1334_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_component *component = codec_dai->component;
+	struct uda1334_priv *uda1334 = snd_soc_component_get_drvdata(component);
+	unsigned int val;
+	int i, j = 0;
+
+	uda1334->sysclk = freq;
+
+	uda1334->rate_constraint.count = 0;
+	for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+		val = freq / lrclk_ratios[i].ratio;
+		/*
+		 * Check that it's a standard rate since core can't
+		 * cope with others and having the odd rates confuses
+		 * constraint matching.
+		 */
+
+		switch (val) {
+		case 8000:
+		case 32000:
+		case 44100:
+		case 48000:
+		case 64000:
+		case 88200:
+		case 96000:
+			dev_dbg(component->dev, "Supported sample rate: %dHz\n",
+				val);
+			uda1334->rate_constraint_list[j++] = val;
+			uda1334->rate_constraint.count++;
+			break;
+		default:
+			dev_dbg(component->dev, "Skipping sample rate: %dHz\n",
+				val);
+		}
+	}
+
+	/* Need at least one supported rate... */
+	if (uda1334->rate_constraint.count == 0)
+		return -EINVAL;
+
+	return 0;
+}
+
+static int uda1334_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK |
+		SND_SOC_DAIFMT_MASTER_MASK);
+
+	if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		    SND_SOC_DAIFMT_CBS_CFS)) {
+		dev_err(codec_dai->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int uda1334_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
+{
+	struct uda1334_priv *uda1334 = snd_soc_component_get_drvdata(dai->component);
+
+	if (uda1334->mute)
+		gpiod_set_value_cansleep(uda1334->mute, mute);
+
+	return 0;
+}
+
+#define UDA1334_RATES SNDRV_PCM_RATE_8000_96000
+
+#define UDA1334_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops uda1334_dai_ops = {
+	.startup	= uda1334_startup,
+	.shutdown	= uda1334_shutdown,
+	.set_sysclk	= uda1334_set_dai_sysclk,
+	.set_fmt	= uda1334_set_fmt,
+	.mute_stream	= uda1334_mute_stream,
+};
+
+static struct snd_soc_dai_driver uda1334_dai = {
+	.name = "uda1334-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = UDA1334_RATES,
+		.formats = UDA1334_FORMATS,
+	},
+	.ops = &uda1334_dai_ops,
+};
+
+static int uda1334_probe(struct snd_soc_component *component)
+{
+	struct uda1334_priv *uda1334 = snd_soc_component_get_drvdata(component);
+
+	uda1334->rate_constraint.list = &uda1334->rate_constraint_list[0];
+	uda1334->rate_constraint.count =
+		ARRAY_SIZE(uda1334->rate_constraint_list);
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver soc_component_dev_uda1334 = {
+	.probe			= uda1334_probe,
+	.controls		= uda1334_snd_controls,
+	.num_controls		= ARRAY_SIZE(uda1334_snd_controls),
+	.dapm_widgets		= uda1334_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(uda1334_dapm_widgets),
+	.dapm_routes		= uda1334_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(uda1334_dapm_routes),
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static const struct of_device_id uda1334_of_match[] = {
+	{ .compatible = "nxp,uda1334" },
+	{ /* sentinel*/ }
+};
+MODULE_DEVICE_TABLE(of, uda1334_of_match);
+
+static int uda1334_codec_probe(struct platform_device *pdev)
+{
+	struct uda1334_priv *uda1334;
+	int ret;
+
+	uda1334 = devm_kzalloc(&pdev->dev, sizeof(struct uda1334_priv),
+			       GFP_KERNEL);
+	if (!uda1334)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, uda1334);
+
+	uda1334->mute = devm_gpiod_get(&pdev->dev, "nxp,mute", GPIOD_OUT_LOW);
+	if (IS_ERR(uda1334->mute)) {
+		ret = PTR_ERR(uda1334->mute);
+		dev_err(&pdev->dev, "Failed to get mute line: %d\n", ret);
+		return ret;
+	}
+
+	uda1334->deemph = devm_gpiod_get(&pdev->dev, "nxp,deemph", GPIOD_OUT_LOW);
+	if (IS_ERR(uda1334->deemph)) {
+		ret = PTR_ERR(uda1334->deemph);
+		dev_err(&pdev->dev, "Failed to get deemph line: %d\n", ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &soc_component_dev_uda1334,
+					      &uda1334_dai, 1);
+	if (ret < 0)
+		dev_err(&pdev->dev, "Failed to register component: %d\n", ret);
+
+	return ret;
+}
+
+static struct platform_driver uda1334_codec_driver = {
+	.probe		= uda1334_codec_probe,
+	.driver		= {
+		.name	= "uda1334-codec",
+		.of_match_table = uda1334_of_match,
+	},
+};
+module_platform_driver(uda1334_codec_driver);
+
+MODULE_DESCRIPTION("ASoC UDA1334 driver");
+MODULE_AUTHOR("Andra Danciu <andradanciu1997@gmail.com>");
+MODULE_ALIAS("platform:uda1334-codec");
+MODULE_LICENSE("GPL v2");
-- 
cgit v1.2.3-59-g8ed1b


From 63643b5902c4bf096b504b0563f5426ba5baef15 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 29 Jul 2019 10:51:47 -0500
Subject: ASoC: Intel: Skylake: move NHLT header to common directory

Prepare move from NHLT code to common directory, starting with header.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/intel-nhlt.h             | 119 +++++++++++++++++++++++++++++++++
 sound/soc/intel/skylake/skl-nhlt.c     |   1 +
 sound/soc/intel/skylake/skl-nhlt.h     | 119 ---------------------------------
 sound/soc/intel/skylake/skl-ssp-clk.c  |   1 +
 sound/soc/intel/skylake/skl-topology.c |   1 +
 sound/soc/intel/skylake/skl.h          |   1 -
 6 files changed, 122 insertions(+), 120 deletions(-)
 create mode 100644 include/sound/intel-nhlt.h
 delete mode 100644 sound/soc/intel/skylake/skl-nhlt.h

(limited to 'sound')

diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h
new file mode 100644
index 000000000000..f85fbf9c7ce4
--- /dev/null
+++ b/include/sound/intel-nhlt.h
@@ -0,0 +1,119 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ *  skl-nhlt.h - Intel HDA Platform NHLT header
+ *
+ *  Copyright (C) 2015 Intel Corp
+ *  Author: Sanjiv Kumar <sanjiv.kumar@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+#ifndef __SKL_NHLT_H__
+#define __SKL_NHLT_H__
+
+#include <linux/acpi.h>
+
+struct wav_fmt {
+	u16 fmt_tag;
+	u16 channels;
+	u32 samples_per_sec;
+	u32 avg_bytes_per_sec;
+	u16 block_align;
+	u16 bits_per_sample;
+	u16 cb_size;
+} __packed;
+
+struct wav_fmt_ext {
+	struct wav_fmt fmt;
+	union samples {
+		u16 valid_bits_per_sample;
+		u16 samples_per_block;
+		u16 reserved;
+	} sample;
+	u32 channel_mask;
+	u8 sub_fmt[16];
+} __packed;
+
+enum nhlt_link_type {
+	NHLT_LINK_HDA = 0,
+	NHLT_LINK_DSP = 1,
+	NHLT_LINK_DMIC = 2,
+	NHLT_LINK_SSP = 3,
+	NHLT_LINK_INVALID
+};
+
+enum nhlt_device_type {
+	NHLT_DEVICE_BT = 0,
+	NHLT_DEVICE_DMIC = 1,
+	NHLT_DEVICE_I2S = 4,
+	NHLT_DEVICE_INVALID
+};
+
+struct nhlt_specific_cfg {
+	u32 size;
+	u8 caps[0];
+} __packed;
+
+struct nhlt_fmt_cfg {
+	struct wav_fmt_ext fmt_ext;
+	struct nhlt_specific_cfg config;
+} __packed;
+
+struct nhlt_fmt {
+	u8 fmt_count;
+	struct nhlt_fmt_cfg fmt_config[0];
+} __packed;
+
+struct nhlt_endpoint {
+	u32  length;
+	u8   linktype;
+	u8   instance_id;
+	u16  vendor_id;
+	u16  device_id;
+	u16  revision_id;
+	u32  subsystem_id;
+	u8   device_type;
+	u8   direction;
+	u8   virtual_bus_id;
+	struct nhlt_specific_cfg config;
+} __packed;
+
+struct nhlt_acpi_table {
+	struct acpi_table_header header;
+	u8 endpoint_count;
+	struct nhlt_endpoint desc[0];
+} __packed;
+
+struct nhlt_resource_desc  {
+	u32 extra;
+	u16 flags;
+	u64 addr_spc_gra;
+	u64 min_addr;
+	u64 max_addr;
+	u64 addr_trans_offset;
+	u64 length;
+} __packed;
+
+#define MIC_ARRAY_2CH 2
+#define MIC_ARRAY_4CH 4
+
+struct nhlt_tdm_config {
+	u8 virtual_slot;
+	u8 config_type;
+} __packed;
+
+struct nhlt_dmic_array_config {
+	struct nhlt_tdm_config tdm_config;
+	u8 array_type;
+} __packed;
+
+enum {
+	NHLT_MIC_ARRAY_2CH_SMALL = 0xa,
+	NHLT_MIC_ARRAY_2CH_BIG = 0xb,
+	NHLT_MIC_ARRAY_4CH_1ST_GEOM = 0xc,
+	NHLT_MIC_ARRAY_4CH_L_SHAPED = 0xd,
+	NHLT_MIC_ARRAY_4CH_2ND_GEOM = 0xe,
+	NHLT_MIC_ARRAY_VENDOR_DEFINED = 0xf,
+};
+
+#endif
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 1132109cb992..aabc5d71650e 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -9,6 +9,7 @@
  * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  */
 #include <linux/pci.h>
+#include <sound/intel-nhlt.h>
 #include "skl.h"
 #include "skl-i2s.h"
 
diff --git a/sound/soc/intel/skylake/skl-nhlt.h b/sound/soc/intel/skylake/skl-nhlt.h
deleted file mode 100644
index f85fbf9c7ce4..000000000000
--- a/sound/soc/intel/skylake/skl-nhlt.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- *  skl-nhlt.h - Intel HDA Platform NHLT header
- *
- *  Copyright (C) 2015 Intel Corp
- *  Author: Sanjiv Kumar <sanjiv.kumar@intel.com>
- *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- */
-#ifndef __SKL_NHLT_H__
-#define __SKL_NHLT_H__
-
-#include <linux/acpi.h>
-
-struct wav_fmt {
-	u16 fmt_tag;
-	u16 channels;
-	u32 samples_per_sec;
-	u32 avg_bytes_per_sec;
-	u16 block_align;
-	u16 bits_per_sample;
-	u16 cb_size;
-} __packed;
-
-struct wav_fmt_ext {
-	struct wav_fmt fmt;
-	union samples {
-		u16 valid_bits_per_sample;
-		u16 samples_per_block;
-		u16 reserved;
-	} sample;
-	u32 channel_mask;
-	u8 sub_fmt[16];
-} __packed;
-
-enum nhlt_link_type {
-	NHLT_LINK_HDA = 0,
-	NHLT_LINK_DSP = 1,
-	NHLT_LINK_DMIC = 2,
-	NHLT_LINK_SSP = 3,
-	NHLT_LINK_INVALID
-};
-
-enum nhlt_device_type {
-	NHLT_DEVICE_BT = 0,
-	NHLT_DEVICE_DMIC = 1,
-	NHLT_DEVICE_I2S = 4,
-	NHLT_DEVICE_INVALID
-};
-
-struct nhlt_specific_cfg {
-	u32 size;
-	u8 caps[0];
-} __packed;
-
-struct nhlt_fmt_cfg {
-	struct wav_fmt_ext fmt_ext;
-	struct nhlt_specific_cfg config;
-} __packed;
-
-struct nhlt_fmt {
-	u8 fmt_count;
-	struct nhlt_fmt_cfg fmt_config[0];
-} __packed;
-
-struct nhlt_endpoint {
-	u32  length;
-	u8   linktype;
-	u8   instance_id;
-	u16  vendor_id;
-	u16  device_id;
-	u16  revision_id;
-	u32  subsystem_id;
-	u8   device_type;
-	u8   direction;
-	u8   virtual_bus_id;
-	struct nhlt_specific_cfg config;
-} __packed;
-
-struct nhlt_acpi_table {
-	struct acpi_table_header header;
-	u8 endpoint_count;
-	struct nhlt_endpoint desc[0];
-} __packed;
-
-struct nhlt_resource_desc  {
-	u32 extra;
-	u16 flags;
-	u64 addr_spc_gra;
-	u64 min_addr;
-	u64 max_addr;
-	u64 addr_trans_offset;
-	u64 length;
-} __packed;
-
-#define MIC_ARRAY_2CH 2
-#define MIC_ARRAY_4CH 4
-
-struct nhlt_tdm_config {
-	u8 virtual_slot;
-	u8 config_type;
-} __packed;
-
-struct nhlt_dmic_array_config {
-	struct nhlt_tdm_config tdm_config;
-	u8 array_type;
-} __packed;
-
-enum {
-	NHLT_MIC_ARRAY_2CH_SMALL = 0xa,
-	NHLT_MIC_ARRAY_2CH_BIG = 0xb,
-	NHLT_MIC_ARRAY_4CH_1ST_GEOM = 0xc,
-	NHLT_MIC_ARRAY_4CH_L_SHAPED = 0xd,
-	NHLT_MIC_ARRAY_4CH_2ND_GEOM = 0xe,
-	NHLT_MIC_ARRAY_VENDOR_DEFINED = 0xf,
-};
-
-#endif
diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c
index 5bb6e40d4d3e..5bfcd46452f9 100644
--- a/sound/soc/intel/skylake/skl-ssp-clk.c
+++ b/sound/soc/intel/skylake/skl-ssp-clk.c
@@ -11,6 +11,7 @@
 #include <linux/platform_device.h>
 #include <linux/clk-provider.h>
 #include <linux/clkdev.h>
+#include <sound/intel-nhlt.h>
 #include "skl.h"
 #include "skl-ssp-clk.h"
 #include "skl-topology.h"
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 6241e35213af..f8a501cf5fbd 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -12,6 +12,7 @@
 #include <linux/types.h>
 #include <linux/firmware.h>
 #include <linux/uuid.h>
+#include <sound/intel-nhlt.h>
 #include <sound/soc.h>
 #include <sound/soc-topology.h>
 #include <uapi/sound/snd_sst_tokens.h>
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 6070666a6392..928e8115a1a7 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -16,7 +16,6 @@
 #include <sound/hdaudio_ext.h>
 #include <sound/hda_codec.h>
 #include <sound/soc.h>
-#include "skl-nhlt.h"
 #include "skl-ssp-clk.h"
 
 #define SKL_SUSPEND_DELAY 2000
-- 
cgit v1.2.3-59-g8ed1b


From 303681f4356d322232dd5f6d9eb4bc62666064c5 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 29 Jul 2019 10:51:48 -0500
Subject: ALSA: hda: move parts of NHLT code to new module

Move parts of the code outside of the Skylake driver to help detect
the presence of DMICs (which are not supported by the HDaudio legacy
driver).

No functionality change (except for the removal of useless OR
operations), only indentation and checkpatch fixes, making sure
that the code compiles without ACPI and fixing an ACPI leak

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/intel-nhlt.h |  41 ++++++++++++++----
 sound/hda/Kconfig          |   5 +++
 sound/hda/Makefile         |   3 ++
 sound/hda/intel-nhlt.c     | 103 +++++++++++++++++++++++++++++++++++++++++++++
 4 files changed, 144 insertions(+), 8 deletions(-)
 create mode 100644 sound/hda/intel-nhlt.c

(limited to 'sound')

diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h
index f85fbf9c7ce4..857922f03931 100644
--- a/include/sound/intel-nhlt.h
+++ b/include/sound/intel-nhlt.h
@@ -1,18 +1,17 @@
 /* SPDX-License-Identifier: GPL-2.0-only */
 /*
- *  skl-nhlt.h - Intel HDA Platform NHLT header
+ *  intel-nhlt.h - Intel HDA Platform NHLT header
  *
- *  Copyright (C) 2015 Intel Corp
- *  Author: Sanjiv Kumar <sanjiv.kumar@intel.com>
- *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *  Copyright (c) 2015-2019 Intel Corporation
  */
-#ifndef __SKL_NHLT_H__
-#define __SKL_NHLT_H__
+
+#ifndef __INTEL_NHLT_H__
+#define __INTEL_NHLT_H__
 
 #include <linux/acpi.h>
 
+#if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_INTEL_NHLT)
+
 struct wav_fmt {
 	u16 fmt_tag;
 	u16 channels;
@@ -116,4 +115,30 @@ enum {
 	NHLT_MIC_ARRAY_VENDOR_DEFINED = 0xf,
 };
 
+struct nhlt_acpi_table *intel_nhlt_init(struct device *dev);
+
+void intel_nhlt_free(struct nhlt_acpi_table *addr);
+
+int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt);
+
+#else
+
+struct nhlt_acpi_table;
+
+static inline struct nhlt_acpi_table *intel_nhlt_init(struct device *dev)
+{
+	return NULL;
+}
+
+static inline void intel_nhlt_free(struct nhlt_acpi_table *addr)
+{
+}
+
+static inline int intel_nhlt_get_dmic_geo(struct device *dev,
+					  struct nhlt_acpi_table *nhlt)
+{
+	return 0;
+}
+#endif
+
 #endif
diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig
index f6feced15f17..9ccbcb5a06bd 100644
--- a/sound/hda/Kconfig
+++ b/sound/hda/Kconfig
@@ -29,3 +29,8 @@ config SND_HDA_PREALLOC_SIZE
 
 	  Note that the pre-allocation size can be changed dynamically
 	  via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too.
+
+config SND_INTEL_NHLT
+	tristate
+	# this config should be selected only for Intel ACPI platforms.
+	# A fallback is provided so that the code compiles in all cases.
\ No newline at end of file
diff --git a/sound/hda/Makefile b/sound/hda/Makefile
index 2160202e2dc1..8560f6ef1b19 100644
--- a/sound/hda/Makefile
+++ b/sound/hda/Makefile
@@ -13,3 +13,6 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o
 
 #extended hda
 obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/
+
+snd-intel-nhlt-objs := intel-nhlt.o
+obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o
diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c
new file mode 100644
index 000000000000..7a62e03ba407
--- /dev/null
+++ b/sound/hda/intel-nhlt.c
@@ -0,0 +1,103 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2015-2019 Intel Corporation
+
+#include <linux/acpi.h>
+#include <sound/intel-nhlt.h>
+
+#define NHLT_ACPI_HEADER_SIG	"NHLT"
+
+/* Unique identification for getting NHLT blobs */
+static guid_t osc_guid =
+	GUID_INIT(0xA69F886E, 0x6CEB, 0x4594,
+		  0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53);
+
+struct nhlt_acpi_table *intel_nhlt_init(struct device *dev)
+{
+	acpi_handle handle;
+	union acpi_object *obj;
+	struct nhlt_resource_desc *nhlt_ptr;
+	struct nhlt_acpi_table *nhlt_table = NULL;
+
+	handle = ACPI_HANDLE(dev);
+	if (!handle) {
+		dev_err(dev, "Didn't find ACPI_HANDLE\n");
+		return NULL;
+	}
+
+	obj = acpi_evaluate_dsm(handle, &osc_guid, 1, 1, NULL);
+
+	if (!obj)
+		return NULL;
+
+	if (obj->type != ACPI_TYPE_BUFFER) {
+		dev_dbg(dev, "No NHLT table found\n");
+		ACPI_FREE(obj);
+		return NULL;
+	}
+
+	nhlt_ptr = (struct nhlt_resource_desc  *)obj->buffer.pointer;
+	if (nhlt_ptr->length)
+		nhlt_table = (struct nhlt_acpi_table *)
+			memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
+				 MEMREMAP_WB);
+	ACPI_FREE(obj);
+	if (nhlt_table &&
+	    (strncmp(nhlt_table->header.signature,
+		     NHLT_ACPI_HEADER_SIG,
+		     strlen(NHLT_ACPI_HEADER_SIG)) != 0)) {
+		memunmap(nhlt_table);
+		dev_err(dev, "NHLT ACPI header signature incorrect\n");
+		return NULL;
+	}
+	return nhlt_table;
+}
+EXPORT_SYMBOL_GPL(intel_nhlt_init);
+
+void intel_nhlt_free(struct nhlt_acpi_table *nhlt)
+{
+	memunmap((void *)nhlt);
+}
+EXPORT_SYMBOL_GPL(intel_nhlt_free);
+
+int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt)
+{
+	struct nhlt_endpoint *epnt;
+	struct nhlt_dmic_array_config *cfg;
+	unsigned int dmic_geo = 0;
+	u8 j;
+
+	if (!nhlt)
+		return 0;
+
+	epnt = (struct nhlt_endpoint *)nhlt->desc;
+
+	for (j = 0; j < nhlt->endpoint_count; j++) {
+		if (epnt->linktype == NHLT_LINK_DMIC) {
+			cfg = (struct nhlt_dmic_array_config  *)
+					(epnt->config.caps);
+			switch (cfg->array_type) {
+			case NHLT_MIC_ARRAY_2CH_SMALL:
+			case NHLT_MIC_ARRAY_2CH_BIG:
+				dmic_geo = MIC_ARRAY_2CH;
+				break;
+
+			case NHLT_MIC_ARRAY_4CH_1ST_GEOM:
+			case NHLT_MIC_ARRAY_4CH_L_SHAPED:
+			case NHLT_MIC_ARRAY_4CH_2ND_GEOM:
+				dmic_geo = MIC_ARRAY_4CH;
+				break;
+
+			default:
+				dev_warn(dev, "undefined DMIC array_type 0x%0x\n",
+					 cfg->array_type);
+			}
+		}
+		epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length);
+	}
+
+	return dmic_geo;
+}
+EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel NHLT driver");
-- 
cgit v1.2.3-59-g8ed1b


From 7a33ea70e1868ee578fe2e9a85dd300efa1a35d5 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 29 Jul 2019 10:51:49 -0500
Subject: ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry

The NHLT spec defines a VENDOR_DEFINED geometry, which requires
reading additional information to figure out the number of
microphones.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/intel-nhlt.h | 10 ++++++++--
 sound/hda/intel-nhlt.c     |  6 +++++-
 2 files changed, 13 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h
index 857922f03931..f657fd8fc0ad 100644
--- a/include/sound/intel-nhlt.h
+++ b/include/sound/intel-nhlt.h
@@ -96,16 +96,22 @@ struct nhlt_resource_desc  {
 #define MIC_ARRAY_2CH 2
 #define MIC_ARRAY_4CH 4
 
-struct nhlt_tdm_config {
+struct nhlt_device_specific_config {
 	u8 virtual_slot;
 	u8 config_type;
 } __packed;
 
 struct nhlt_dmic_array_config {
-	struct nhlt_tdm_config tdm_config;
+	struct nhlt_device_specific_config device_config;
 	u8 array_type;
 } __packed;
 
+struct nhlt_vendor_dmic_array_config {
+	struct nhlt_dmic_array_config dmic_config;
+	u8 nb_mics;
+	/* TODO add vendor mic config */
+} __packed;
+
 enum {
 	NHLT_MIC_ARRAY_2CH_SMALL = 0xa,
 	NHLT_MIC_ARRAY_2CH_BIG = 0xb,
diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c
index 7a62e03ba407..daede96f28ee 100644
--- a/sound/hda/intel-nhlt.c
+++ b/sound/hda/intel-nhlt.c
@@ -63,6 +63,7 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt)
 {
 	struct nhlt_endpoint *epnt;
 	struct nhlt_dmic_array_config *cfg;
+	struct nhlt_vendor_dmic_array_config *cfg_vendor;
 	unsigned int dmic_geo = 0;
 	u8 j;
 
@@ -86,7 +87,10 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt)
 			case NHLT_MIC_ARRAY_4CH_2ND_GEOM:
 				dmic_geo = MIC_ARRAY_4CH;
 				break;
-
+			case NHLT_MIC_ARRAY_VENDOR_DEFINED:
+				cfg_vendor = (struct nhlt_vendor_dmic_array_config *)cfg;
+				dmic_geo = cfg_vendor->nb_mics;
+				break;
 			default:
 				dev_warn(dev, "undefined DMIC array_type 0x%0x\n",
 					 cfg->array_type);
-- 
cgit v1.2.3-59-g8ed1b


From 1169cbf6b98e11397715d457e42070e066325fc0 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 29 Jul 2019 10:51:50 -0500
Subject: ASoC: Intel: Skylake: use common NHLT module

No functionality change, only use common functions now.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/intel/Kconfig            |  1 +
 sound/soc/intel/skylake/skl-nhlt.c | 90 --------------------------------------
 sound/soc/intel/skylake/skl.c      | 12 +++--
 sound/soc/intel/skylake/skl.h      |  3 --
 4 files changed, 9 insertions(+), 97 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 96a00a9d4cf8..a3ec17fd63cd 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -215,6 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON
 	select SND_SOC_INTEL_SST
 	select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
 	select SND_SOC_ACPI_INTEL_MATCH
+	select SND_INTEL_NHLT
 	help
 	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
 	  GeminiLake or CannonLake platform with the DSP enabled in the BIOS
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index aabc5d71650e..6f57ceb9efb7 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -13,54 +13,6 @@
 #include "skl.h"
 #include "skl-i2s.h"
 
-#define NHLT_ACPI_HEADER_SIG	"NHLT"
-
-/* Unique identification for getting NHLT blobs */
-static guid_t osc_guid =
-	GUID_INIT(0xA69F886E, 0x6CEB, 0x4594,
-		  0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53);
-
-
-struct nhlt_acpi_table *skl_nhlt_init(struct device *dev)
-{
-	acpi_handle handle;
-	union acpi_object *obj;
-	struct nhlt_resource_desc  *nhlt_ptr = NULL;
-	struct nhlt_acpi_table *nhlt_table = NULL;
-
-	handle = ACPI_HANDLE(dev);
-	if (!handle) {
-		dev_err(dev, "Didn't find ACPI_HANDLE\n");
-		return NULL;
-	}
-
-	obj = acpi_evaluate_dsm(handle, &osc_guid, 1, 1, NULL);
-	if (obj && obj->type == ACPI_TYPE_BUFFER) {
-		nhlt_ptr = (struct nhlt_resource_desc  *)obj->buffer.pointer;
-		if (nhlt_ptr->length)
-			nhlt_table = (struct nhlt_acpi_table *)
-				memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
-				MEMREMAP_WB);
-		ACPI_FREE(obj);
-		if (nhlt_table && (strncmp(nhlt_table->header.signature,
-					NHLT_ACPI_HEADER_SIG,
-					strlen(NHLT_ACPI_HEADER_SIG)) != 0)) {
-			memunmap(nhlt_table);
-			dev_err(dev, "NHLT ACPI header signature incorrect\n");
-			return NULL;
-		}
-		return nhlt_table;
-	}
-
-	dev_err(dev, "device specific method to extract NHLT blob failed\n");
-	return NULL;
-}
-
-void skl_nhlt_free(struct nhlt_acpi_table *nhlt)
-{
-	memunmap((void *) nhlt);
-}
-
 static struct nhlt_specific_cfg *skl_get_specific_cfg(
 		struct device *dev, struct nhlt_fmt *fmt,
 		u8 no_ch, u32 rate, u16 bps, u8 linktype)
@@ -163,48 +115,6 @@ struct nhlt_specific_cfg
 	return NULL;
 }
 
-int skl_get_dmic_geo(struct skl *skl)
-{
-	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
-	struct nhlt_endpoint *epnt;
-	struct nhlt_dmic_array_config *cfg;
-	struct device *dev = &skl->pci->dev;
-	unsigned int dmic_geo = 0;
-	u8 j;
-
-	if (!nhlt)
-		return 0;
-
-	epnt = (struct nhlt_endpoint *)nhlt->desc;
-
-	for (j = 0; j < nhlt->endpoint_count; j++) {
-		if (epnt->linktype == NHLT_LINK_DMIC) {
-			cfg = (struct nhlt_dmic_array_config  *)
-					(epnt->config.caps);
-			switch (cfg->array_type) {
-			case NHLT_MIC_ARRAY_2CH_SMALL:
-			case NHLT_MIC_ARRAY_2CH_BIG:
-				dmic_geo |= MIC_ARRAY_2CH;
-				break;
-
-			case NHLT_MIC_ARRAY_4CH_1ST_GEOM:
-			case NHLT_MIC_ARRAY_4CH_L_SHAPED:
-			case NHLT_MIC_ARRAY_4CH_2ND_GEOM:
-				dmic_geo |= MIC_ARRAY_4CH;
-				break;
-
-			default:
-				dev_warn(dev, "undefined DMIC array_type 0x%0x\n",
-						cfg->array_type);
-
-			}
-		}
-		epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length);
-	}
-
-	return dmic_geo;
-}
-
 static void skl_nhlt_trim_space(char *trim)
 {
 	char *s = trim;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 3362e71b4563..2b5159890a57 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -26,9 +26,11 @@
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
 #include <sound/hda_codec.h>
+#include <sound/intel-nhlt.h>
 #include "skl.h"
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
+
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 #include "../../../soc/codecs/hdac_hda.h"
 #endif
@@ -516,7 +518,9 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 
 	if (pdata) {
 		skl->use_tplg_pcm = pdata->use_tplg_pcm;
-		mach->mach_params.dmic_num = skl_get_dmic_geo(skl);
+		mach->mach_params.dmic_num =
+			intel_nhlt_get_dmic_geo(&skl->pci->dev,
+						skl->nhlt);
 	}
 
 	return 0;
@@ -1029,7 +1033,7 @@ static int skl_probe(struct pci_dev *pci,
 
 	device_disable_async_suspend(bus->dev);
 
-	skl->nhlt = skl_nhlt_init(bus->dev);
+	skl->nhlt = intel_nhlt_init(bus->dev);
 
 	if (skl->nhlt == NULL) {
 #if !IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
@@ -1095,7 +1099,7 @@ out_dsp_free:
 out_clk_free:
 	skl_clock_device_unregister(skl);
 out_nhlt_free:
-	skl_nhlt_free(skl->nhlt);
+	intel_nhlt_free(skl->nhlt);
 out_free:
 	skl_free(bus);
 
@@ -1144,7 +1148,7 @@ static void skl_remove(struct pci_dev *pci)
 	skl_dmic_device_unregister(skl);
 	skl_clock_device_unregister(skl);
 	skl_nhlt_remove_sysfs(skl);
-	skl_nhlt_free(skl->nhlt);
+	intel_nhlt_free(skl->nhlt);
 	skl_free(bus);
 	dev_set_drvdata(&pci->dev, NULL);
 }
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 928e8115a1a7..f4dd6c767993 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -128,13 +128,10 @@ struct skl_dsp_ops {
 int skl_platform_unregister(struct device *dev);
 int skl_platform_register(struct device *dev);
 
-struct nhlt_acpi_table *skl_nhlt_init(struct device *dev);
-void skl_nhlt_free(struct nhlt_acpi_table *addr);
 struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance,
 					u8 link_type, u8 s_fmt, u8 no_ch,
 					u32 s_rate, u8 dirn, u8 dev_type);
 
-int skl_get_dmic_geo(struct skl *skl);
 int skl_nhlt_update_topology_bin(struct skl *skl);
 int skl_init_dsp(struct skl *skl);
 int skl_free_dsp(struct skl *skl);
-- 
cgit v1.2.3-59-g8ed1b


From 8c57588313805d4e12a3d470dee3405c2236655f Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 29 Jul 2019 10:51:51 -0500
Subject: ALSA: hda/intel: stop probe if DMICS are detected on Skylake+
 platforms

The legacy HD-Audio driver cannot handle Skylake+ platforms with
digital microphones. For those platforms, the SOF or SST drivers need
to be used.

This patch provides an automatic way of detecting the presence of
DMICs using NHTL information reported by the BIOS. A kernel kconfig
option or a kernel module parameter provide an opt-in means of
stopping the probe. The kernel would then look for an alternate driver
registered for the same PCI ID to probe.

With this capability, distros no longer have to blacklist
snd-hda-intel, but still need to make sure the SOF/SST drivers are
functional by providing the relevant firmware and topology files in
/lib/firmware/intel

The coexistence between SOF and SST drivers and their dynamic
detection is not addressed by this patch, different mechanisms need to
be used, e.g. DMI-based quirks.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig     | 10 ++++++++++
 sound/pci/hda/hda_intel.c | 34 ++++++++++++++++++++++++++++++++++
 2 files changed, 44 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 35d934309cb2..b5966014b5f7 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -12,6 +12,7 @@ config SND_HDA_INTEL
 	tristate "HD Audio PCI"
 	depends on SND_PCI
 	select SND_HDA
+	select SND_INTEL_NHLT if ACPI
 	help
 	  Say Y here to include support for Intel "High Definition
 	  Audio" (Azalia) and its compatible devices.
@@ -22,6 +23,15 @@ config SND_HDA_INTEL
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-hda-intel.
 
+config SND_HDA_INTEL_DETECT_DMIC
+	bool "DMIC detection and probe abort"
+	depends on SND_HDA_INTEL
+	help
+	  Say Y to detect digital microphones on SKL+ devices. DMICs
+	  cannot be handled by the HDaudio legacy driver and are
+	  currently only supported by the SOF driver.
+	  If unsure say N.
+
 config SND_HDA_TEGRA
 	tristate "NVIDIA Tegra HD Audio"
 	depends on ARCH_TEGRA
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cb8b0945547c..ae9c13248a1d 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -46,6 +46,7 @@
 #include <sound/initval.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
+#include <sound/intel-nhlt.h>
 #include <linux/vgaarb.h>
 #include <linux/vga_switcheroo.h>
 #include <linux/firmware.h>
@@ -124,6 +125,7 @@ static char *patch[SNDRV_CARDS];
 static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] =
 					CONFIG_SND_HDA_INPUT_BEEP_MODE};
 #endif
+static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC);
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
@@ -158,6 +160,8 @@ module_param_array(beep_mode, bool, NULL, 0444);
 MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode "
 			    "(0=off, 1=on) (default=1).");
 #endif
+module_param(dmic_detect, bool, 0444);
+MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms");
 
 #ifdef CONFIG_PM
 static int param_set_xint(const char *val, const struct kernel_param *kp);
@@ -2025,6 +2029,25 @@ static const struct hda_controller_ops pci_hda_ops = {
 	.position_check = azx_position_check,
 };
 
+static int azx_check_dmic(struct pci_dev *pci, struct azx *chip)
+{
+	struct nhlt_acpi_table *nhlt;
+	int ret = 0;
+
+	if (chip->driver_type == AZX_DRIVER_SKL &&
+	    pci->class != 0x040300) {
+		nhlt = intel_nhlt_init(&pci->dev);
+		if (nhlt) {
+			if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) {
+				ret = -ENODEV;
+				dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n");
+			}
+			intel_nhlt_free(nhlt);
+		}
+	}
+	return ret;
+}
+
 static int azx_probe(struct pci_dev *pci,
 		     const struct pci_device_id *pci_id)
 {
@@ -2055,6 +2078,17 @@ static int azx_probe(struct pci_dev *pci,
 	card->private_data = chip;
 	hda = container_of(chip, struct hda_intel, chip);
 
+	/*
+	 * stop probe if digital microphones detected on Skylake+ platform
+	 * with the DSP enabled. This is an opt-in behavior defined at build
+	 * time or at run-time with a module parameter
+	 */
+	if (dmic_detect) {
+		err = azx_check_dmic(pci, chip);
+		if (err < 0)
+			goto out_free;
+	}
+
 	pci_set_drvdata(pci, card);
 
 	err = register_vga_switcheroo(chip);
-- 
cgit v1.2.3-59-g8ed1b


From a342031cdd0818cb0fbcb44798211c7a02c7ca27 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:47 +0200
Subject: ASoC: create pcm for codec2codec links as well

At the moment, codec to codec links uses an ephemeral variable for
the struct snd_pcm_substream. Also the struct snd_soc_pcm_runtime
does not have real struct snd_pcm.

This might a problem if the functions used by a codec on codec to
codec link expect these structures to exist, and keep on existing
during the life of the codec.

For example, it is the case of the hdmi-codec, which uses
snd_pcm_add_chmap_ctls(). For the controls to works, the pcm and
substream must to exist.

This change is first step, it create pcm (and substreams) for codec
to codec links, in the same way as dpcm backend links.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 42 ++++++++++++------------------------------
 sound/soc/soc-pcm.c  | 35 ++++++++++++++++++++++++++++++++---
 2 files changed, 44 insertions(+), 33 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 06697b2d96b1..da11e44b01aa 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -447,16 +447,6 @@ static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card)
 		flush_delayed_work(&rtd->delayed_work);
 }
 
-static void codec2codec_close_delayed_work(struct work_struct *work)
-{
-	/*
-	 * Currently nothing to do for c2c links
-	 * Since c2c links are internal nodes in the DAPM graph and
-	 * don't interface with the outside world or application layer
-	 * we don't have to do any special handling on close.
-	 */
-}
-
 #ifdef CONFIG_PM_SLEEP
 /* powers down audio subsystem for suspend */
 int snd_soc_suspend(struct device *dev)
@@ -1555,27 +1545,19 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		return ret;
 	}
 
-	if (!dai_link->params) {
-		/* create the pcm */
-		ret = soc_new_pcm(rtd, num);
-		if (ret < 0) {
-			dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
-				dai_link->stream_name, ret);
-			return ret;
-		}
-		ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
-		if (ret < 0)
-			return ret;
-		ret = soc_link_dai_pcm_new(rtd->codec_dais,
-					   rtd->num_codecs, rtd);
-		if (ret < 0)
-			return ret;
-	} else {
-		INIT_DELAYED_WORK(&rtd->delayed_work,
-				  codec2codec_close_delayed_work);
+	/* create the pcm */
+	ret = soc_new_pcm(rtd, num);
+	if (ret < 0) {
+		dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
+			dai_link->stream_name, ret);
+		return ret;
 	}
-
-	return 0;
+	ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
+	if (ret < 0)
+		return ret;
+	ret = soc_link_dai_pcm_new(rtd->codec_dais,
+				   rtd->num_codecs, rtd);
+	return ret;
 }
 
 static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fabeac164a6c..30264bc592f6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -678,6 +678,16 @@ static void close_delayed_work(struct work_struct *work)
 	mutex_unlock(&rtd->pcm_mutex);
 }
 
+static void codec2codec_close_delayed_work(struct work_struct *work)
+{
+	/*
+	 * Currently nothing to do for c2c links
+	 * Since c2c links are internal nodes in the DAPM graph and
+	 * don't interface with the outside world or application layer
+	 * we don't have to do any special handling on close.
+	 */
+}
+
 /*
  * Called by ALSA when a PCM substream is closed. Private data can be
  * freed here. The cpu DAI, codec DAI, machine and components are also
@@ -3011,6 +3021,12 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		playback = rtd->dai_link->dpcm_playback;
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
+		/* Adapt stream for codec2codec links */
+		struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ?
+			&cpu_dai->driver->playback : &cpu_dai->driver->capture;
+		struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ?
+			&cpu_dai->driver->capture : &cpu_dai->driver->playback;
+
 		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
 			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK))
@@ -3019,6 +3035,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_CAPTURE))
 				capture = 1;
 		}
+
+		capture = capture && cpu_capture->channels_min;
+		playback = playback && cpu_playback->channels_min;
 	}
 
 	if (rtd->dai_link->playback_only) {
@@ -3032,7 +3051,13 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	}
 
 	/* create the PCM */
-	if (rtd->dai_link->no_pcm) {
+	if (rtd->dai_link->params) {
+		snprintf(new_name, sizeof(new_name), "codec2codec(%s)",
+			 rtd->dai_link->stream_name);
+
+		ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
+					   playback, capture, &pcm);
+	} else if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",
 			rtd->dai_link->stream_name);
 
@@ -3059,13 +3084,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n",num, new_name);
 
 	/* DAPM dai link stream work */
-	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+	if (rtd->dai_link->params)
+		INIT_DELAYED_WORK(&rtd->delayed_work,
+				  codec2codec_close_delayed_work);
+	else
+		INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
 	pcm->nonatomic = rtd->dai_link->nonatomic;
 	rtd->pcm = pcm;
 	pcm->private_data = rtd;
 
-	if (rtd->dai_link->no_pcm) {
+	if (rtd->dai_link->no_pcm || rtd->dai_link->params) {
 		if (playback)
 			pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
 		if (capture)
-- 
cgit v1.2.3-59-g8ed1b


From a72706ed8208ac3f72d1c3ebbc6509e368b0dcb0 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:48 +0200
Subject: ASoC: codec2codec: remove ephemeral variables

Now that codec to codec links struct snd_soc_pcm_runtime have lasting pcm
and substreams, let's use them. Alsa allocate and keep the
struct snd_pcm_runtime as long as the link is powered.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-6-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 72 ++++++++++++++++++++++++++++++----------------------
 1 file changed, 42 insertions(+), 30 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 2d183e2d23de..1c953a1b46ce 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3775,6 +3775,7 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_hw_params *params = NULL;
 	const struct snd_soc_pcm_stream *config = NULL;
+	struct snd_pcm_runtime *runtime = NULL;
 	unsigned int fmt;
 	int ret = 0;
 
@@ -3782,6 +3783,14 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	if (!params)
 		return -ENOMEM;
 
+	runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
+	if (!runtime) {
+		ret = -ENOMEM;
+		goto out;
+	}
+
+	substream->runtime = runtime;
+
 	substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 	snd_soc_dapm_widget_for_each_source_path(w, path) {
 		source = path->source->priv;
@@ -3808,6 +3817,8 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 		sink->active++;
 	}
 
+	substream->hw_opened = 1;
+
 	/*
 	 * Note: getting the config after .startup() gives a chance to
 	 * either party on the link to alter the configuration if
@@ -3864,6 +3875,9 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 	}
 
 out:
+	if (ret < 0)
+		kfree(runtime);
+
 	kfree(params);
 	return ret;
 }
@@ -3873,29 +3887,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
-	struct snd_soc_pcm_runtime *rtd = w->priv;
-	struct snd_pcm_substream substream;
-	struct snd_pcm_runtime *runtime = NULL;
-	int ret = 0;
+	struct snd_pcm_substream *substream = w->priv;
+	int ret = 0, saved_stream = substream->stream;
 
 	if (WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
 		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
 		return -EINVAL;
 
-	memset(&substream, 0, sizeof(substream));
-
-	/* Allocate a dummy snd_pcm_runtime for startup() and other ops() */
-	runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
-	if (!runtime) {
-		ret = -ENOMEM;
-		goto out;
-	}
-	substream.runtime = runtime;
-	substream.private_data = rtd;
-
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		ret = snd_soc_dai_link_event_pre_pmu(w, &substream);
+		ret = snd_soc_dai_link_event_pre_pmu(w, substream);
 		if (ret < 0)
 			goto out;
 
@@ -3926,40 +3927,45 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 			ret = 0;
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
-			snd_soc_dai_hw_free(source, &substream);
+			snd_soc_dai_hw_free(source, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
-			snd_soc_dai_hw_free(sink, &substream);
+			snd_soc_dai_hw_free(sink, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		substream->stream = SNDRV_PCM_STREAM_CAPTURE;
 		snd_soc_dapm_widget_for_each_source_path(w, path) {
 			source = path->source->priv;
 			source->active--;
-			snd_soc_dai_shutdown(source, &substream);
+			snd_soc_dai_shutdown(source, substream);
 		}
 
-		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		substream->stream = SNDRV_PCM_STREAM_PLAYBACK;
 		snd_soc_dapm_widget_for_each_sink_path(w, path) {
 			sink = path->sink->priv;
 			sink->active--;
-			snd_soc_dai_shutdown(sink, &substream);
+			snd_soc_dai_shutdown(sink, substream);
 		}
 		break;
 
+	case SND_SOC_DAPM_POST_PMD:
+		kfree(substream->runtime);
+		break;
+
 	default:
 		WARN(1, "Unknown event %d\n", event);
 		ret = -EINVAL;
 	}
 
 out:
-	kfree(runtime);
+	/* Restore the substream direction */
+	substream->stream = saved_stream;
 	return ret;
 }
 
@@ -4082,9 +4088,11 @@ outfree_w_param:
 }
 
 static struct snd_soc_dapm_widget *
-snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
+snd_soc_dapm_new_dai(struct snd_soc_card *card,
+		     struct snd_pcm_substream *substream,
 		     char *id)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
 	const char **w_param_text;
@@ -4103,7 +4111,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 	template.name = link_name;
 	template.event = snd_soc_dai_link_event;
 	template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		SND_SOC_DAPM_PRE_PMD;
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD;
 	template.kcontrol_news = NULL;
 
 	/* allocate memory for control, only in case of multiple configs */
@@ -4138,7 +4146,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
 		goto outfree_kcontrol_news;
 	}
 
-	w->priv = rtd;
+	w->priv = substream;
 
 	return w;
 
@@ -4260,6 +4268,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
 	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
+	struct snd_pcm_substream *substream;
+	struct snd_pcm_str *streams = rtd->pcm->streams;
 	int i;
 
 	if (rtd->dai_link->params) {
@@ -4278,7 +4288,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 
 		if (playback_cpu && codec) {
 			if (!playback) {
-				playback = snd_soc_dapm_new_dai(card, rtd,
+				substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+				playback = snd_soc_dapm_new_dai(card, substream,
 								"playback");
 				if (IS_ERR(playback)) {
 					dev_err(rtd->dev,
@@ -4307,7 +4318,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 
 		if (codec && capture_cpu) {
 			if (!capture) {
-				capture = snd_soc_dapm_new_dai(card, rtd,
+				substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+				capture = snd_soc_dapm_new_dai(card, substream,
 							       "capture");
 				if (IS_ERR(capture)) {
 					dev_err(rtd->dev,
-- 
cgit v1.2.3-59-g8ed1b


From 9de98628c895d15427138073986eab1e3ce39cb4 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 25 Jul 2019 18:59:49 +0200
Subject: ASoC: codec2codec: fill some of the runtime stream parameters

Set the information provided struct snd_soc_pcm_stream in the
struct snd_pcm_runtime of the codec to codec link.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190725165949.29699-7-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1c953a1b46ce..e16838e1bda2 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3874,6 +3874,11 @@ snd_soc_dai_link_event_pre_pmu(struct snd_soc_dapm_widget *w,
 		dapm_update_dai_unlocked(substream, params, sink);
 	}
 
+	runtime->format = params_format(params);
+	runtime->subformat = params_subformat(params);
+	runtime->channels = params_channels(params);
+	runtime->rate = params_rate(params);
+
 out:
 	if (ret < 0)
 		kfree(runtime);
-- 
cgit v1.2.3-59-g8ed1b


From d8701fae95c0e6cb587b293cb8e01acc96dd1095 Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Wed, 31 Jul 2019 15:17:14 +0200
Subject: ASoC: Intel: SoC: skylake: no need to check return value of
 debugfs_create functions

When calling debugfs functions, there is no need to ever check the
return value.  The function can work or not, but the code logic should
never do something different based on this.

Also, if a debugfs call fails, userspace is notified with an error in
the log, so no need to log the error again.

Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: Jie Yang <yang.jie@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20190731131716.9764-1-gregkh@linuxfoundation.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-debug.c | 28 +++++-----------------------
 1 file changed, 5 insertions(+), 23 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index fb232428109f..212370bf704c 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -164,10 +164,8 @@ void skl_debug_init_module(struct skl_debug *d,
 			struct snd_soc_dapm_widget *w,
 			struct skl_module_cfg *mconfig)
 {
-	if (!debugfs_create_file(w->name, 0444,
-				d->modules, mconfig,
-				&mcfg_fops))
-		dev_err(d->dev, "%s: module debugfs init failed\n", w->name);
+	debugfs_create_file(w->name, 0444, d->modules, mconfig,
+			    &mcfg_fops);
 }
 
 static ssize_t fw_softreg_read(struct file *file, char __user *user_buf,
@@ -224,34 +222,18 @@ struct skl_debug *skl_debugfs_init(struct skl_dev *skl)
 		return NULL;
 
 	/* create the debugfs dir with platform component's debugfs as parent */
-	d->fs = debugfs_create_dir("dsp",
-				   skl->component->debugfs_root);
-	if (IS_ERR(d->fs) || !d->fs) {
-		dev_err(&skl->pci->dev, "debugfs root creation failed\n");
-		return NULL;
-	}
+	d->fs = debugfs_create_dir("dsp", skl->component->debugfs_root);
 
 	d->skl = skl;
 	d->dev = &skl->pci->dev;
 
 	/* now create the module dir */
 	d->modules = debugfs_create_dir("modules", d->fs);
-	if (IS_ERR(d->modules) || !d->modules) {
-		dev_err(&skl->pci->dev, "modules debugfs create failed\n");
-		goto err;
-	}
 
-	if (!debugfs_create_file("fw_soft_regs_rd", 0444, d->fs, d,
-				 &soft_regs_ctrl_fops)) {
-		dev_err(d->dev, "fw soft regs control debugfs init failed\n");
-		goto err;
-	}
+	debugfs_create_file("fw_soft_regs_rd", 0444, d->fs, d,
+			    &soft_regs_ctrl_fops);
 
 	return d;
-
-err:
-	debugfs_remove_recursive(d->fs);
-	return NULL;
 }
 
 void skl_debugfs_exit(struct skl_dev *skl)
-- 
cgit v1.2.3-59-g8ed1b


From fee531d6fc49b9a616525e30955c4cf3b403f632 Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Wed, 31 Jul 2019 15:17:15 +0200
Subject: ASoC: core: no need to check return value of debugfs_create functions

When calling debugfs functions, there is no need to ever check the
return value.  The function can work or not, but the code logic should
never do something different based on this.

Also, there is no need to store the individual debugfs file name, just
remove the whole directory all at once, saving a local variable.

Note, the soc-pcm "state" file has now moved to a subdirectory, as it is
only a good idea to save the dentries for debugfs directories, not
individual files, as the individual file debugfs functions are changing
to not return a dentry.

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20190731131716.9764-2-gregkh@linuxfoundation.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  1 -
 sound/soc/soc-core.c | 43 ++++++-------------------------------------
 sound/soc/soc-dapm.c | 30 ++++--------------------------
 sound/soc/soc-pcm.c  | 14 ++++----------
 4 files changed, 14 insertions(+), 74 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index d770606732cd..f0797792dd8d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1220,7 +1220,6 @@ struct snd_soc_card {
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_card_root;
-	struct dentry *debugfs_pop_time;
 #endif
 	u32 pop_time;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index da11e44b01aa..ed66d2c68d10 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -165,13 +165,6 @@ static void soc_init_component_debugfs(struct snd_soc_component *component)
 				component->card->debugfs_card_root);
 	}
 
-	if (IS_ERR(component->debugfs_root)) {
-		dev_warn(component->dev,
-			"ASoC: Failed to create component debugfs directory: %ld\n",
-			PTR_ERR(component->debugfs_root));
-		return;
-	}
-
 	snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component),
 		component->debugfs_root);
 }
@@ -215,32 +208,15 @@ DEFINE_SHOW_ATTRIBUTE(component_list);
 
 static void soc_init_card_debugfs(struct snd_soc_card *card)
 {
-	if (!snd_soc_debugfs_root)
-		return;
-
 	card->debugfs_card_root = debugfs_create_dir(card->name,
 						     snd_soc_debugfs_root);
-	if (IS_ERR(card->debugfs_card_root)) {
-		dev_warn(card->dev,
-			 "ASoC: Failed to create card debugfs directory: %ld\n",
-			 PTR_ERR(card->debugfs_card_root));
-		card->debugfs_card_root = NULL;
-		return;
-	}
 
-	card->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
-						    card->debugfs_card_root,
-						    &card->pop_time);
-	if (IS_ERR(card->debugfs_pop_time))
-		dev_warn(card->dev,
-			 "ASoC: Failed to create pop time debugfs file: %ld\n",
-			 PTR_ERR(card->debugfs_pop_time));
+	debugfs_create_u32("dapm_pop_time", 0644, card->debugfs_card_root,
+			   &card->pop_time);
 }
 
 static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
 {
-	if (!card->debugfs_card_root)
-		return;
 	debugfs_remove_recursive(card->debugfs_card_root);
 	card->debugfs_card_root = NULL;
 }
@@ -248,19 +224,12 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
 static void snd_soc_debugfs_init(void)
 {
 	snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL);
-	if (IS_ERR_OR_NULL(snd_soc_debugfs_root)) {
-		pr_warn("ASoC: Failed to create debugfs directory\n");
-		snd_soc_debugfs_root = NULL;
-		return;
-	}
 
-	if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL,
-				 &dai_list_fops))
-		pr_warn("ASoC: Failed to create DAI list debugfs file\n");
+	debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL,
+			    &dai_list_fops);
 
-	if (!debugfs_create_file("components", 0444, snd_soc_debugfs_root, NULL,
-				 &component_list_fops))
-		pr_warn("ASoC: Failed to create component list debugfs file\n");
+	debugfs_create_file("components", 0444, snd_soc_debugfs_root, NULL,
+			    &component_list_fops);
 }
 
 static void snd_soc_debugfs_exit(void)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index e16838e1bda2..d93c1038fab0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2154,50 +2154,28 @@ static const struct file_operations dapm_bias_fops = {
 void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
 	struct dentry *parent)
 {
-	struct dentry *d;
-
 	if (!parent || IS_ERR(parent))
 		return;
 
 	dapm->debugfs_dapm = debugfs_create_dir("dapm", parent);
 
-	if (IS_ERR(dapm->debugfs_dapm)) {
-		dev_warn(dapm->dev,
-			 "ASoC: Failed to create DAPM debugfs directory %ld\n",
-			 PTR_ERR(dapm->debugfs_dapm));
-		return;
-	}
-
-	d = debugfs_create_file("bias_level", 0444,
-				dapm->debugfs_dapm, dapm,
-				&dapm_bias_fops);
-	if (IS_ERR(d))
-		dev_warn(dapm->dev,
-			 "ASoC: Failed to create bias level debugfs file: %ld\n",
-			 PTR_ERR(d));
+	debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm,
+			    &dapm_bias_fops);
 }
 
 static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
 {
 	struct snd_soc_dapm_context *dapm = w->dapm;
-	struct dentry *d;
 
 	if (!dapm->debugfs_dapm || !w->name)
 		return;
 
-	d = debugfs_create_file(w->name, 0444,
-				dapm->debugfs_dapm, w,
-				&dapm_widget_power_fops);
-	if (IS_ERR(d))
-		dev_warn(w->dapm->dev,
-			 "ASoC: Failed to create %s debugfs file: %ld\n",
-			 w->name, PTR_ERR(d));
+	debugfs_create_file(w->name, 0444, dapm->debugfs_dapm, w,
+			    &dapm_widget_power_fops);
 }
 
 static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm)
 {
-	if (!dapm->debugfs_dapm)
-		return;
 	debugfs_remove_recursive(dapm->debugfs_dapm);
 	dapm->debugfs_dapm = NULL;
 }
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 30264bc592f6..ce7297c37537 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1205,9 +1205,9 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
 			stream ? "<-" : "->", be->dai_link->name);
 
 #ifdef CONFIG_DEBUG_FS
-	if (fe->debugfs_dpcm_root)
-		dpcm->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644,
-				fe->debugfs_dpcm_root, &dpcm->state);
+	dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name,
+						 fe->debugfs_dpcm_root);
+	debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state);
 #endif
 	return 1;
 }
@@ -1262,7 +1262,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
 		dpcm_be_reparent(fe, dpcm->be, stream);
 
 #ifdef CONFIG_DEBUG_FS
-		debugfs_remove(dpcm->debugfs_state);
+		debugfs_remove_recursive(dpcm->debugfs_state);
 #endif
 		spin_lock_irqsave(&fe->card->dpcm_lock, flags);
 		list_del(&dpcm->list_be);
@@ -3415,12 +3415,6 @@ void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
 
 	rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name,
 			rtd->card->debugfs_card_root);
-	if (!rtd->debugfs_dpcm_root) {
-		dev_dbg(rtd->dev,
-			 "ASoC: Failed to create dpcm debugfs directory %s\n",
-			 rtd->dai_link->name);
-		return;
-	}
 
 	debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root,
 			    rtd, &dpcm_state_fops);
-- 
cgit v1.2.3-59-g8ed1b


From 3ff3a4f657b3fab2d56247983c2ebed180ef2fbb Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Date: Wed, 31 Jul 2019 15:17:16 +0200
Subject: ASoC: SOF: no need to check return value of debugfs_create functions

When calling debugfs functions, there is no need to ever check the
return value.  The function can work or not, but the code logic should
never do something different based on this.

Also, if a debugfs call fails, userspace is notified with an error in
the log, so no need to log the error again.

Because we no longer need to check the return value, there's no need to
save the dentry returned by debugfs.  Just use the dentry in the file
pointer if we really need to figure out the "name" of the file being
opened.

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20190731131716.9764-3-gregkh@linuxfoundation.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/debug.c    | 49 ++++++++++++++++++------------------------------
 sound/soc/sof/sof-priv.h |  1 -
 sound/soc/sof/trace.c    |  9 ++-------
 3 files changed, 20 insertions(+), 39 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index 2388477a965e..40940b2fe9d5 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -128,6 +128,7 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer,
 	unsigned long ipc_duration_ms = 0;
 	bool flood_duration_test = false;
 	unsigned long ipc_count = 0;
+	struct dentry *dentry;
 	int err;
 #endif
 	size_t size;
@@ -149,11 +150,12 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer,
 	 * ipc_duration_ms test floods the DSP for the time specified
 	 * in the debugfs entry.
 	 */
-	if (strcmp(dfse->dfsentry->d_name.name, "ipc_flood_count") &&
-	    strcmp(dfse->dfsentry->d_name.name, "ipc_flood_duration_ms"))
+	dentry = file->f_path.dentry;
+	if (strcmp(dentry->d_name.name, "ipc_flood_count") &&
+	    strcmp(dentry->d_name.name, "ipc_flood_duration_ms"))
 		return -EINVAL;
 
-	if (!strcmp(dfse->dfsentry->d_name.name, "ipc_flood_duration_ms"))
+	if (!strcmp(dentry->d_name.name, "ipc_flood_duration_ms"))
 		flood_duration_test = true;
 
 	/* test completion criterion */
@@ -219,6 +221,7 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer,
 {
 	struct snd_sof_dfsentry *dfse = file->private_data;
 	struct snd_sof_dev *sdev = dfse->sdev;
+	struct dentry *dentry;
 	loff_t pos = *ppos;
 	size_t size_ret;
 	int skip = 0;
@@ -226,8 +229,9 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer,
 	u8 *buf;
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
-	if ((!strcmp(dfse->dfsentry->d_name.name, "ipc_flood_count") ||
-	     !strcmp(dfse->dfsentry->d_name.name, "ipc_flood_duration_ms")) &&
+	dentry = file->f_path.dentry;
+	if ((!strcmp(dentry->d_name.name, "ipc_flood_count") ||
+	     !strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) &&
 	    dfse->cache_buf) {
 		if (*ppos)
 			return 0;
@@ -290,8 +294,7 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer,
 		if (!pm_runtime_active(sdev->dev) &&
 		    dfse->access_type == SOF_DEBUGFS_ACCESS_D0_ONLY) {
 			dev_err(sdev->dev,
-				"error: debugfs entry %s cannot be read in DSP D3\n",
-				dfse->dfsentry->d_name.name);
+				"error: debugfs entry cannot be read in DSP D3\n");
 			kfree(buf);
 			return -EINVAL;
 		}
@@ -356,17 +359,11 @@ int snd_sof_debugfs_io_item(struct snd_sof_dev *sdev,
 	}
 #endif
 
-	dfse->dfsentry = debugfs_create_file(name, 0444, sdev->debugfs_root,
-					     dfse, &sof_dfs_fops);
-	if (!dfse->dfsentry) {
-		/* can't rely on debugfs, only log error and keep going */
-		dev_err(sdev->dev, "error: cannot create debugfs entry %s\n",
-			name);
-	} else {
-		/* add to dfsentry list */
-		list_add(&dfse->list, &sdev->dfsentry_list);
+	debugfs_create_file(name, 0444, sdev->debugfs_root, dfse,
+			    &sof_dfs_fops);
 
-	}
+	/* add to dfsentry list */
+	list_add(&dfse->list, &sdev->dfsentry_list);
 
 	return 0;
 }
@@ -402,16 +399,10 @@ int snd_sof_debugfs_buf_item(struct snd_sof_dev *sdev,
 		return -ENOMEM;
 #endif
 
-	dfse->dfsentry = debugfs_create_file(name, mode, sdev->debugfs_root,
-					     dfse, &sof_dfs_fops);
-	if (!dfse->dfsentry) {
-		/* can't rely on debugfs, only log error and keep going */
-		dev_err(sdev->dev, "error: cannot create debugfs entry %s\n",
-			name);
-	} else {
-		/* add to dfsentry list */
-		list_add(&dfse->list, &sdev->dfsentry_list);
-	}
+	debugfs_create_file(name, mode, sdev->debugfs_root, dfse,
+			    &sof_dfs_fops);
+	/* add to dfsentry list */
+	list_add(&dfse->list, &sdev->dfsentry_list);
 
 	return 0;
 }
@@ -426,10 +417,6 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev)
 
 	/* use "sof" as top level debugFS dir */
 	sdev->debugfs_root = debugfs_create_dir("sof", NULL);
-	if (IS_ERR_OR_NULL(sdev->debugfs_root)) {
-		dev_err(sdev->dev, "error: failed to create debugfs directory\n");
-		return 0;
-	}
 
 	/* init dfsentry list */
 	INIT_LIST_HEAD(&sdev->dfsentry_list);
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index 983eadef4b30..1cec3f23f9cd 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -230,7 +230,6 @@ enum sof_debugfs_access_type {
 
 /* FS entry for debug files that can expose DSP memories, registers */
 struct snd_sof_dfsentry {
-	struct dentry *dfsentry;
 	size_t size;
 	enum sof_dfsentry_type type;
 	/*
diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c
index befed975161c..4c3cff031fd6 100644
--- a/sound/soc/sof/trace.c
+++ b/sound/soc/sof/trace.c
@@ -148,13 +148,8 @@ static int trace_debugfs_create(struct snd_sof_dev *sdev)
 	dfse->size = sdev->dmatb.bytes;
 	dfse->sdev = sdev;
 
-	dfse->dfsentry = debugfs_create_file("trace", 0444, sdev->debugfs_root,
-					     dfse, &sof_dfs_trace_fops);
-	if (!dfse->dfsentry) {
-		/* can't rely on debugfs, only log error and keep going */
-		dev_err(sdev->dev,
-			"error: cannot create debugfs entry for trace\n");
-	}
+	debugfs_create_file("trace", 0444, sdev->debugfs_root, dfse,
+			    &sof_dfs_trace_fops);
 
 	return 0;
 }
-- 
cgit v1.2.3-59-g8ed1b


From f24e41d3d04f326613d8a7ebecf72c3019826f71 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Wed, 31 Jul 2019 23:32:34 +0100
Subject: ASoC: zx-tdm: remove redundant assignment to ts_width on error return
 path

The value assigned to ts_width is never read on the error return path
so the assignment is redundant and can be removed.  Remove it.

Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190731223234.16153-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/zte/zx-tdm.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c
index 5e877fe9ba7b..0e5a05b25a77 100644
--- a/sound/soc/zte/zx-tdm.c
+++ b/sound/soc/zte/zx-tdm.c
@@ -211,7 +211,6 @@ static int zx_tdm_hw_params(struct snd_pcm_substream *substream,
 		ts_width = 1;
 		break;
 	default:
-		ts_width = 0;
 		dev_err(socdai->dev, "Unknown data format\n");
 		return -EINVAL;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From 3bb4aefd2aed628452b8ce544b76abda0ce531f0 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Thu, 1 Aug 2019 17:28:24 +0100
Subject: ALSA: isa/wavefront: remove redundant assignment to pointer bptr

The pointer bptr is being assigned a value that is never read
and it is being updated in the next statement with a new value.
The initialization is redundant and can be removed.

Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/isa/wavefront/wavefront_synth.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index aec1c46e6697..c5b1d5900eed 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -788,7 +788,6 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header)
 
 	dev->patch_status[header->number] |= WF_SLOT_FILLED;
 
-	bptr = buf;
 	bptr = munge_int32 (header->number, buf, 2);
 	munge_buf ((unsigned char *)&header->hdr.p, bptr, WF_PATCH_BYTES);
     
-- 
cgit v1.2.3-59-g8ed1b


From b6fcab14ff027c4d54505d9cdff41535c3daf077 Mon Sep 17 00:00:00 2001
From: Tony W Wang-oc <TonyWWang-oc@zhaoxin.com>
Date: Fri, 2 Aug 2019 03:04:08 +0000
Subject: ALSA: hda: Add support of Zhaoxin controller

Add the new PCI ID 0x1d17 0x3288 Zhaoxin controller support

Signed-off-by: Tony W Wang-oc <TonyWWang-oc@zhaoxin.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 635d971b003b..151f206e1c02 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -270,6 +270,7 @@ enum {
 	AZX_DRIVER_CTX,
 	AZX_DRIVER_CTHDA,
 	AZX_DRIVER_CMEDIA,
+	AZX_DRIVER_ZHAOXIN,
 	AZX_DRIVER_GENERIC,
 	AZX_NUM_DRIVERS, /* keep this as last entry */
 };
@@ -384,6 +385,7 @@ static char *driver_short_names[] = {
 	[AZX_DRIVER_CTX] = "HDA Creative", 
 	[AZX_DRIVER_CTHDA] = "HDA Creative",
 	[AZX_DRIVER_CMEDIA] = "HDA C-Media",
+	[AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin",
 	[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
 };
 
@@ -2624,6 +2626,8 @@ static const struct pci_device_id azx_ids[] = {
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
+	/* Zhaoxin */
+	{ PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
-- 
cgit v1.2.3-59-g8ed1b


From cf9441adb1a35506d7606866c382b9d8614169b5 Mon Sep 17 00:00:00 2001
From: Stephen Boyd <swboyd@chromium.org>
Date: Tue, 30 Jul 2019 11:15:49 -0700
Subject: ASoC: Remove dev_err() usage after platform_get_irq()

We don't need dev_err() messages when platform_get_irq() fails now that
platform_get_irq() prints an error message itself when something goes
wrong. Let's remove these prints with a simple semantic patch.

// <smpl>
@@
expression ret;
struct platform_device *E;
@@

ret =
(
platform_get_irq(E, ...)
|
platform_get_irq_byname(E, ...)
);

if ( \( ret < 0 \| ret <= 0 \) )
{
(
-if (ret != -EPROBE_DEFER)
-{ ...
-dev_err(...);
-... }
|
...
-dev_err(...);
)
...
}
// </smpl>

While we're here, remove braces on if statements that only have one
statement (manually).

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Link: https://lore.kernel.org/r/20190730181557.90391-50-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/atmel-classd.c             |  7 ++-----
 sound/soc/atmel/atmel-pdmic.c              |  7 ++-----
 sound/soc/bcm/cygnus-ssp.c                 |  7 ++-----
 sound/soc/codecs/msm8916-wcd-analog.c      | 12 +++---------
 sound/soc/codecs/twl6040.c                 |  4 +---
 sound/soc/fsl/fsl_asrc.c                   |  4 +---
 sound/soc/fsl/fsl_esai.c                   |  4 +---
 sound/soc/fsl/fsl_sai.c                    |  4 +---
 sound/soc/fsl/fsl_spdif.c                  |  4 +---
 sound/soc/fsl/fsl_ssi.c                    |  4 +---
 sound/soc/fsl/imx-ssi.c                    |  4 +---
 sound/soc/kirkwood/kirkwood-i2s.c          |  4 +---
 sound/soc/mediatek/common/mtk-btcvsd.c     |  4 +---
 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c |  4 +---
 sound/soc/mediatek/mt8173/mt8173-afe-pcm.c |  4 +---
 sound/soc/mxs/mxs-saif.c                   |  8 ++------
 sound/soc/qcom/lpass-platform.c            |  5 +----
 sound/soc/sof/intel/bdw.c                  |  5 +----
 sound/soc/sof/intel/byt.c                  |  5 +----
 sound/soc/sprd/sprd-mcdt.c                 |  4 +---
 sound/soc/sti/sti_uniperif.c               |  4 +---
 sound/soc/stm/stm32_i2s.c                  |  5 +----
 sound/soc/stm/stm32_sai.c                  |  4 +---
 sound/soc/stm/stm32_spdifrx.c              |  4 +---
 sound/soc/sunxi/sun4i-i2s.c                |  4 +---
 sound/soc/uniphier/aio-dma.c               |  4 +---
 sound/soc/xilinx/xlnx_formatter_pcm.c      |  2 --
 sound/soc/xtensa/xtfpga-i2s.c              |  1 -
 28 files changed, 32 insertions(+), 100 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index 0f2c574f27f1..e98601eccfa3 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -571,11 +571,8 @@ static int atmel_classd_probe(struct platform_device *pdev)
 	dd->pdata = pdata;
 
 	dd->irq = platform_get_irq(pdev, 0);
-	if (dd->irq < 0) {
-		ret = dd->irq;
-		dev_err(dev, "failed to could not get irq: %d\n", ret);
-		return ret;
-	}
+	if (dd->irq < 0)
+		return dd->irq;
 
 	dd->pclk = devm_clk_get(dev, "pclk");
 	if (IS_ERR(dd->pclk)) {
diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c
index e09c28349e0d..04ec6f0af179 100644
--- a/sound/soc/atmel/atmel-pdmic.c
+++ b/sound/soc/atmel/atmel-pdmic.c
@@ -612,11 +612,8 @@ static int atmel_pdmic_probe(struct platform_device *pdev)
 	dd->dev = dev;
 
 	dd->irq = platform_get_irq(pdev, 0);
-	if (dd->irq < 0) {
-		ret = dd->irq;
-		dev_err(dev, "failed to get irq: %d\n", ret);
-		return ret;
-	}
+	if (dd->irq < 0)
+		return dd->irq;
 
 	dd->pclk = devm_clk_get(dev, "pclk");
 	if (IS_ERR(dd->pclk)) {
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index b7c358b48d8d..2f9357d7da96 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1342,11 +1342,8 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 	}
 
 	cygaud->irq_num = platform_get_irq(pdev, 0);
-	if (cygaud->irq_num <= 0) {
-		dev_err(dev, "platform_get_irq failed\n");
-		err = cygaud->irq_num;
-		return err;
-	}
+	if (cygaud->irq_num <= 0)
+		return cygaud->irq_num;
 
 	err = audio_clk_init(pdev, cygaud);
 	if (err) {
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index 368b6c09474b..667e9f73aba3 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -1185,10 +1185,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 	}
 
 	irq = platform_get_irq_byname(pdev, "mbhc_switch_int");
-	if (irq < 0) {
-		dev_err(dev, "failed to get mbhc switch irq\n");
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_threaded_irq(dev, irq, NULL,
 			       pm8916_mbhc_switch_irq_handler,
@@ -1200,10 +1198,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 
 	if (priv->mbhc_btn_enabled) {
 		irq = platform_get_irq_byname(pdev, "mbhc_but_press_det");
-		if (irq < 0) {
-			dev_err(dev, "failed to get button press irq\n");
+		if (irq < 0)
 			return irq;
-		}
 
 		ret = devm_request_threaded_irq(dev, irq, NULL,
 				       mbhc_btn_press_irq_handler,
@@ -1214,10 +1210,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 			dev_err(dev, "cannot request mbhc button press irq\n");
 
 		irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det");
-		if (irq < 0) {
-			dev_err(dev, "failed to get button release irq\n");
+		if (irq < 0)
 			return irq;
-		}
 
 		ret = devm_request_threaded_irq(dev, irq, NULL,
 				       mbhc_btn_release_irq_handler,
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 472c2fff34a8..f34637afee51 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1108,10 +1108,8 @@ static int twl6040_probe(struct snd_soc_component *component)
 	priv->component = component;
 
 	priv->plug_irq = platform_get_irq(pdev, 0);
-	if (priv->plug_irq < 0) {
-		dev_err(component->dev, "invalid irq: %d\n", priv->plug_irq);
+	if (priv->plug_irq < 0)
 		return priv->plug_irq;
-	}
 
 	INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work);
 
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index cbbf6257f08a..cfa40ef6b1ca 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -885,10 +885,8 @@ static int fsl_asrc_probe(struct platform_device *pdev)
 	}
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0,
 			       dev_name(&pdev->dev), asrc_priv);
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 4ce8ac769244..5832144beb9f 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -962,10 +962,8 @@ static int fsl_esai_probe(struct platform_device *pdev)
 				PTR_ERR(esai_priv->spbaclk));
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0,
 			       esai_priv->name, esai_priv);
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 6d3c6c8d50ce..8f4d9fa95599 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -831,10 +831,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	}
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai);
 	if (ret) {
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 4842e6df9a2d..7858a5499ac5 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1248,10 +1248,8 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	}
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
 			       dev_name(&pdev->dev), spdif_priv);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index fa862af25c1a..b0a6fead1a6a 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1510,10 +1510,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi->irq = platform_get_irq(pdev, 0);
-	if (ssi->irq < 0) {
-		dev_err(dev, "no irq for node %s\n", pdev->name);
+	if (ssi->irq < 0)
 		return ssi->irq;
-	}
 
 	/* Set software limitations for synchronous mode except AC97 */
 	if (ssi->synchronous && !fsl_ssi_is_ac97(ssi)) {
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 9038b61317be..42031ba7da31 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -520,10 +520,8 @@ static int imx_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi->irq = platform_get_irq(pdev, 0);
-	if (ssi->irq < 0) {
-		dev_err(&pdev->dev, "Failed to get IRQ: %d\n", ssi->irq);
+	if (ssi->irq < 0)
 		return ssi->irq;
-	}
 
 	ssi->clk = devm_clk_get(&pdev->dev, NULL);
 	if (IS_ERR(ssi->clk)) {
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 5076ec4cc7a6..61226fefe1c4 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -537,10 +537,8 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return PTR_ERR(priv->io);
 
 	priv->irq = platform_get_irq(pdev, 0);
-	if (priv->irq < 0) {
-		dev_err(&pdev->dev, "platform_get_irq failed: %d\n", priv->irq);
+	if (priv->irq < 0)
 		return priv->irq;
-	}
 
 	if (np) {
 		priv->burst = 128;		/* might be 32 or 128 */
diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c
index c7a81c4be068..d00608c73c6e 100644
--- a/sound/soc/mediatek/common/mtk-btcvsd.c
+++ b/sound/soc/mediatek/common/mtk-btcvsd.c
@@ -1335,10 +1335,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev)
 
 	/* irq */
 	irq_id = platform_get_irq(pdev, 0);
-	if (irq_id <= 0) {
-		dev_err(dev, "%pOFn no irq found\n", dev->of_node);
+	if (irq_id <= 0)
 		return irq_id < 0 ? irq_id : -ENXIO;
-	}
 
 	ret = devm_request_irq(dev, irq_id, mtk_btcvsd_snd_irq_handler,
 			       IRQF_TRIGGER_LOW, "BTCVSD_ISR_Handle",
diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
index 7064a9fd6f74..9af76ae315a5 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
@@ -1342,10 +1342,8 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	irq_id = platform_get_irq_byname(pdev, "asys");
-	if (irq_id < 0) {
-		dev_err(dev, "unable to get ASYS IRQ\n");
+	if (irq_id < 0)
 		return irq_id;
-	}
 
 	ret = devm_request_irq(dev, irq_id, mt2701_asys_isr,
 			       IRQF_TRIGGER_NONE, "asys-isr", (void *)afe);
diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
index 90bd2c92cae7..0ee29255e731 100644
--- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
+++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
@@ -1074,10 +1074,8 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev)
 	afe->dev = &pdev->dev;
 
 	irq_id = platform_get_irq(pdev, 0);
-	if (irq_id <= 0) {
-		dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node);
+	if (irq_id <= 0)
 		return irq_id < 0 ? irq_id : -ENXIO;
-	}
 	ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler,
 			       0, "Afe_ISR_Handle", (void *)afe);
 	if (ret) {
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index a2c79426513b..1e38ce858326 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -790,12 +790,8 @@ static int mxs_saif_probe(struct platform_device *pdev)
 		return PTR_ERR(saif->base);
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		ret = irq;
-		dev_err(&pdev->dev, "failed to get irq resource: %d\n",
-			ret);
-		return ret;
-	}
+	if (irq < 0)
+		return irq;
 
 	saif->dev = &pdev->dev;
 	ret = devm_request_irq(&pdev->dev, irq, mxs_saif_irq, 0,
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index cf7a299f4547..4c745baa39f7 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -564,11 +564,8 @@ int asoc_qcom_lpass_platform_register(struct platform_device *pdev)
 	int ret;
 
 	drvdata->lpaif_irq = platform_get_irq_byname(pdev, "lpass-irq-lpaif");
-	if (drvdata->lpaif_irq < 0) {
-		dev_err(&pdev->dev, "error getting irq handle: %d\n",
-			drvdata->lpaif_irq);
+	if (drvdata->lpaif_irq < 0)
 		return -ENODEV;
-	}
 
 	/* ensure audio hardware is disabled */
 	ret = regmap_write(drvdata->lpaif_map,
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 70d524ef9bc0..4bb9636da990 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -613,11 +613,8 @@ static int bdw_probe(struct snd_sof_dev *sdev)
 
 	/* register our IRQ */
 	sdev->ipc_irq = platform_get_irq(pdev, desc->irqindex_host_ipc);
-	if (sdev->ipc_irq < 0) {
-		dev_err(sdev->dev, "error: failed to get IRQ at index %d\n",
-			desc->irqindex_host_ipc);
+	if (sdev->ipc_irq < 0)
 		return sdev->ipc_irq;
-	}
 
 	dev_dbg(sdev->dev, "using IRQ %d\n", sdev->ipc_irq);
 	ret = devm_request_threaded_irq(sdev->dev, sdev->ipc_irq,
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 107d711efc3f..000d576f6a8d 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -728,11 +728,8 @@ static int byt_acpi_probe(struct snd_sof_dev *sdev)
 irq:
 	/* register our IRQ */
 	sdev->ipc_irq = platform_get_irq(pdev, desc->irqindex_host_ipc);
-	if (sdev->ipc_irq < 0) {
-		dev_err(sdev->dev, "error: failed to get IRQ at index %d\n",
-			desc->irqindex_host_ipc);
+	if (sdev->ipc_irq < 0)
 		return sdev->ipc_irq;
-	}
 
 	dev_dbg(sdev->dev, "using IRQ %d\n", sdev->ipc_irq);
 	ret = devm_request_threaded_irq(sdev->dev, sdev->ipc_irq,
diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c
index 7448015a4935..f439e5503a3c 100644
--- a/sound/soc/sprd/sprd-mcdt.c
+++ b/sound/soc/sprd/sprd-mcdt.c
@@ -959,10 +959,8 @@ static int sprd_mcdt_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, mcdt);
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_err(&pdev->dev, "Failed to get MCDT interrupt\n");
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_irq(&pdev->dev, irq, sprd_mcdt_irq_handler,
 			       0, "sprd-mcdt", mcdt);
diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c
index 645bcbe91601..ee4a0151e63e 100644
--- a/sound/soc/sti/sti_uniperif.c
+++ b/sound/soc/sti/sti_uniperif.c
@@ -426,10 +426,8 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 				     UNIPERIF_FIFO_DATA_OFFSET(uni);
 
 	uni->irq = platform_get_irq(priv->pdev, 0);
-	if (uni->irq < 0) {
-		dev_err(dev, "Failed to get IRQ resource\n");
+	if (uni->irq < 0)
 		return -ENXIO;
-	}
 
 	uni->type = dev_data->type;
 
diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c
index ba6452dab69b..3e7226a53e53 100644
--- a/sound/soc/stm/stm32_i2s.c
+++ b/sound/soc/stm/stm32_i2s.c
@@ -855,11 +855,8 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev,
 
 	/* Get irqs */
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		if (irq != -EPROBE_DEFER)
-			dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, IRQF_ONESHOT,
 			       dev_name(&pdev->dev), i2s);
diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index 1ac5103cea78..ef4273361d0d 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -193,10 +193,8 @@ static int stm32_sai_probe(struct platform_device *pdev)
 
 	/* init irqs */
 	sai->irq = platform_get_irq(pdev, 0);
-	if (sai->irq < 0) {
-		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+	if (sai->irq < 0)
 		return sai->irq;
-	}
 
 	/* reset */
 	rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c
index ee71b898897b..cd4b235fce57 100644
--- a/sound/soc/stm/stm32_spdifrx.c
+++ b/sound/soc/stm/stm32_spdifrx.c
@@ -909,10 +909,8 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev,
 	}
 
 	spdifrx->irq = platform_get_irq(pdev, 0);
-	if (spdifrx->irq < 0) {
-		dev_err(&pdev->dev, "No irq for node %s\n", pdev->name);
+	if (spdifrx->irq < 0)
 		return spdifrx->irq;
-	}
 
 	return 0;
 }
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9b2232908b65..d97d694c48df 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1087,10 +1087,8 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 		return PTR_ERR(regs);
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_err(&pdev->dev, "Can't retrieve our interrupt\n");
+	if (irq < 0)
 		return irq;
-	}
 
 	i2s->variant = of_device_get_match_data(&pdev->dev);
 	if (!i2s->variant) {
diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c
index 862346d66774..e8446cc4e8f8 100644
--- a/sound/soc/uniphier/aio-dma.c
+++ b/sound/soc/uniphier/aio-dma.c
@@ -289,10 +289,8 @@ int uniphier_aiodma_soc_register_platform(struct platform_device *pdev)
 		return PTR_ERR(chip->regmap);
 
 	irq = platform_get_irq(pdev, 0);
-	if (irq < 0) {
-		dev_err(dev, "Could not get irq.\n");
+	if (irq < 0)
 		return irq;
-	}
 
 	ret = devm_request_irq(dev, irq, aiodma_irq,
 			       IRQF_SHARED, dev_name(dev), pdev);
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index dc8721f4f56b..48970efe7838 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -613,7 +613,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev)
 		aud_drv_data->mm2s_irq = platform_get_irq_byname(pdev,
 								 "irq_mm2s");
 		if (aud_drv_data->mm2s_irq < 0) {
-			dev_err(dev, "xlnx audio mm2s irq resource failed\n");
 			ret = aud_drv_data->mm2s_irq;
 			goto clk_err;
 		}
@@ -640,7 +639,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev)
 		aud_drv_data->s2mm_irq = platform_get_irq_byname(pdev,
 								 "irq_s2mm");
 		if (aud_drv_data->s2mm_irq < 0) {
-			dev_err(dev, "xlnx audio s2mm irq resource failed\n");
 			ret = aud_drv_data->s2mm_irq;
 			goto clk_err;
 		}
diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c
index 9da395d14a8d..efd374f114a0 100644
--- a/sound/soc/xtensa/xtfpga-i2s.c
+++ b/sound/soc/xtensa/xtfpga-i2s.c
@@ -570,7 +570,6 @@ static int xtfpga_i2s_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0) {
-		dev_err(&pdev->dev, "No IRQ resource\n");
 		err = irq;
 		goto err;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From 18bd7b5ef3a313a9af1314508bca1ed982c05f75 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:49:23 +0900
Subject: ASoC: soc-pcm: remove soc_rtdcom_ack()

No ALSA SoC driver has .ack at component->driver->ops.
We can revive it if some-driver want to use it, but let's remove it
so far to avoid maintaining complex code

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87muh15roz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-pcm.c | 22 ----------------------
 1 file changed, 22 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index ce7297c37537..bf04b69d6878 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2873,26 +2873,6 @@ static void soc_pcm_private_free(struct snd_pcm *pcm)
 	}
 }
 
-static int soc_rtdcom_ack(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->ack)
-			continue;
-
-		/* FIXME. it returns 1st ask now */
-		return component->driver->ops->ack(substream);
-	}
-
-	return -EINVAL;
-}
-
 static int soc_rtdcom_copy_user(struct snd_pcm_substream *substream, int channel,
 				unsigned long pos, void __user *buf,
 				unsigned long bytes)
@@ -3129,8 +3109,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		if (!ops)
 			continue;
 
-		if (ops->ack)
-			rtd->ops.ack		= soc_rtdcom_ack;
 		if (ops->copy_user)
 			rtd->ops.copy_user	= soc_rtdcom_copy_user;
 		if (ops->copy_kernel)
-- 
cgit v1.2.3-59-g8ed1b


From 4efbb20971af527990f3c51056d142e3f7dc6b4a Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:49:29 +0900
Subject: ASoC: soc-pcm: remove soc_rtdcom_copy_kernel()

No ALSA SoC driver has .copy_kernel at component->driver->ops.
We can revive it if some-driver want to use it, but let's remove it
so far to avoid maintaining complex code

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lfwl5rot.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-pcm.c | 24 ------------------------
 1 file changed, 24 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index bf04b69d6878..e85548244b55 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2896,28 +2896,6 @@ static int soc_rtdcom_copy_user(struct snd_pcm_substream *substream, int channel
 	return -EINVAL;
 }
 
-static int soc_rtdcom_copy_kernel(struct snd_pcm_substream *substream, int channel,
-				  unsigned long pos, void *buf, unsigned long bytes)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->copy_kernel)
-			continue;
-
-		/* FIXME. it returns 1st copy now */
-		return component->driver->ops->copy_kernel(substream, channel,
-							   pos, buf, bytes);
-	}
-
-	return -EINVAL;
-}
-
 static int soc_rtdcom_fill_silence(struct snd_pcm_substream *substream, int channel,
 				   unsigned long pos, unsigned long bytes)
 {
@@ -3111,8 +3089,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 
 		if (ops->copy_user)
 			rtd->ops.copy_user	= soc_rtdcom_copy_user;
-		if (ops->copy_kernel)
-			rtd->ops.copy_kernel	= soc_rtdcom_copy_kernel;
 		if (ops->fill_silence)
 			rtd->ops.fill_silence	= soc_rtdcom_fill_silence;
 		if (ops->page)
-- 
cgit v1.2.3-59-g8ed1b


From 57cb28856ed8db11faa7cc1c751eb6ad00e7c3f0 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:49:43 +0900
Subject: ASoC: soc-pcm: remove soc_fill_silence()

No ALSA SoC driver has .fill_silence at component->driver->ops.
We can revive it if some-driver want to use it, but let's remove it
so far to avoid maintaining complex code

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k1c55rof.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-pcm.c | 24 ------------------------
 1 file changed, 24 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e85548244b55..1e6c4e226933 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2896,28 +2896,6 @@ static int soc_rtdcom_copy_user(struct snd_pcm_substream *substream, int channel
 	return -EINVAL;
 }
 
-static int soc_rtdcom_fill_silence(struct snd_pcm_substream *substream, int channel,
-				   unsigned long pos, unsigned long bytes)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->fill_silence)
-			continue;
-
-		/* FIXME. it returns 1st silence now */
-		return component->driver->ops->fill_silence(substream, channel,
-							    pos, bytes);
-	}
-
-	return -EINVAL;
-}
-
 static struct page *soc_rtdcom_page(struct snd_pcm_substream *substream,
 				    unsigned long offset)
 {
@@ -3089,8 +3067,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 
 		if (ops->copy_user)
 			rtd->ops.copy_user	= soc_rtdcom_copy_user;
-		if (ops->fill_silence)
-			rtd->ops.fill_silence	= soc_rtdcom_fill_silence;
 		if (ops->page)
 			rtd->ops.page		= soc_rtdcom_page;
 		if (ops->mmap)
-- 
cgit v1.2.3-59-g8ed1b


From 4ff1fef10f353b928bcc9d56d31fda53f2c43191 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:49:48 +0900
Subject: ASoC: add soc-component.c

ALSA SoC has many snd_soc_component_xxx(), but these are randomly
located in many files. Because of it, code is difficult to read.
This patch creates new soc-component.c, and moves existing
snd_soc_component_xxx() into it.
But not yet fully. We need more cleanup it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87imrp5roa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h | 328 ++++++++++++++++++++++++++++++++++++++++++
 include/sound/soc.h           | 306 +--------------------------------------
 sound/soc/Makefile            |   2 +-
 sound/soc/soc-component.c     | 269 ++++++++++++++++++++++++++++++++++
 sound/soc/soc-core.c          |  44 ------
 sound/soc/soc-jack.c          |  18 ---
 sound/soc/soc-utils.c         | 199 -------------------------
 7 files changed, 599 insertions(+), 567 deletions(-)
 create mode 100644 include/sound/soc-component.h
 create mode 100644 sound/soc/soc-component.c

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
new file mode 100644
index 000000000000..a97d499e5d7a
--- /dev/null
+++ b/include/sound/soc-component.h
@@ -0,0 +1,328 @@
+/* SPDX-License-Identifier: GPL-2.0
+ *
+ * soc-component.h
+ *
+ * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#ifndef __SOC_COMPONENT_H
+#define __SOC_COMPONENT_H
+
+#include <sound/soc.h>
+
+/*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST	-2
+#define SND_SOC_COMP_ORDER_EARLY	-1
+#define SND_SOC_COMP_ORDER_NORMAL	 0
+#define SND_SOC_COMP_ORDER_LATE		 1
+#define SND_SOC_COMP_ORDER_LAST		 2
+
+#define for_each_comp_order(order)		\
+	for (order  = SND_SOC_COMP_ORDER_FIRST;	\
+	     order <= SND_SOC_COMP_ORDER_LAST;	\
+	     order++)
+
+/* component interface */
+struct snd_soc_component_driver {
+	const char *name;
+
+	/* Default control and setup, added after probe() is run */
+	const struct snd_kcontrol_new *controls;
+	unsigned int num_controls;
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	unsigned int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	unsigned int num_dapm_routes;
+
+	int (*probe)(struct snd_soc_component *component);
+	void (*remove)(struct snd_soc_component *component);
+	int (*suspend)(struct snd_soc_component *component);
+	int (*resume)(struct snd_soc_component *component);
+
+	unsigned int (*read)(struct snd_soc_component *component,
+			     unsigned int reg);
+	int (*write)(struct snd_soc_component *component,
+		     unsigned int reg, unsigned int val);
+
+	/* pcm creation and destruction */
+	int (*pcm_new)(struct snd_soc_pcm_runtime *rtd);
+	void (*pcm_free)(struct snd_pcm *pcm);
+
+	/* component wide operations */
+	int (*set_sysclk)(struct snd_soc_component *component,
+			  int clk_id, int source, unsigned int freq, int dir);
+	int (*set_pll)(struct snd_soc_component *component, int pll_id,
+		       int source, unsigned int freq_in, unsigned int freq_out);
+	int (*set_jack)(struct snd_soc_component *component,
+			struct snd_soc_jack *jack,  void *data);
+
+	/* DT */
+	int (*of_xlate_dai_name)(struct snd_soc_component *component,
+				 struct of_phandle_args *args,
+				 const char **dai_name);
+	int (*of_xlate_dai_id)(struct snd_soc_component *comment,
+			       struct device_node *endpoint);
+	void (*seq_notifier)(struct snd_soc_component *component,
+			     enum snd_soc_dapm_type type, int subseq);
+	int (*stream_event)(struct snd_soc_component *component, int event);
+	int (*set_bias_level)(struct snd_soc_component *component,
+			      enum snd_soc_bias_level level);
+
+	const struct snd_pcm_ops *ops;
+	const struct snd_compr_ops *compr_ops;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
+
+	/*
+	 * signal if the module handling the component should not be removed
+	 * if a pcm is open. Setting this would prevent the module
+	 * refcount being incremented in probe() but allow it be incremented
+	 * when a pcm is opened and decremented when it is closed.
+	 */
+	unsigned int module_get_upon_open:1;
+
+	/* bits */
+	unsigned int idle_bias_on:1;
+	unsigned int suspend_bias_off:1;
+	unsigned int use_pmdown_time:1; /* care pmdown_time at stop */
+	unsigned int endianness:1;
+	unsigned int non_legacy_dai_naming:1;
+
+	/* this component uses topology and ignore machine driver FEs */
+	const char *ignore_machine;
+	const char *topology_name_prefix;
+	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
+				  struct snd_pcm_hw_params *params);
+	bool use_dai_pcm_id;	/* use DAI link PCM ID as PCM device number */
+	int be_pcm_base;	/* base device ID for all BE PCMs */
+};
+
+struct snd_soc_component {
+	const char *name;
+	int id;
+	const char *name_prefix;
+	struct device *dev;
+	struct snd_soc_card *card;
+
+	unsigned int active;
+
+	unsigned int suspended:1; /* is in suspend PM state */
+
+	struct list_head list;
+	struct list_head card_aux_list; /* for auxiliary bound components */
+	struct list_head card_list;
+
+	const struct snd_soc_component_driver *driver;
+
+	struct list_head dai_list;
+	int num_dai;
+
+	struct regmap *regmap;
+	int val_bytes;
+
+	struct mutex io_mutex;
+
+	/* attached dynamic objects */
+	struct list_head dobj_list;
+
+	/*
+	 * DO NOT use any of the fields below in drivers, they are temporary and
+	 * are going to be removed again soon. If you use them in driver code
+	 * the driver will be marked as BROKEN when these fields are removed.
+	 */
+
+	/* Don't use these, use snd_soc_component_get_dapm() */
+	struct snd_soc_dapm_context dapm;
+
+	/* machine specific init */
+	int (*init)(struct snd_soc_component *component);
+
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_root;
+	const char *debugfs_prefix;
+#endif
+};
+
+#define for_each_component_dais(component, dai)\
+	list_for_each_entry(dai, &(component)->dai_list, list)
+#define for_each_component_dais_safe(component, dai, _dai)\
+	list_for_each_entry_safe(dai, _dai, &(component)->dai_list, list)
+
+/**
+ * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is
+ *  embedded in
+ * @dapm: The DAPM context to cast to the component
+ *
+ * This function must only be used on DAPM contexts that are known to be part of
+ * a component (e.g. in a component driver). Otherwise the behavior is
+ * undefined.
+ */
+static inline struct snd_soc_component *snd_soc_dapm_to_component(
+	struct snd_soc_dapm_context *dapm)
+{
+	return container_of(dapm, struct snd_soc_component, dapm);
+}
+
+/**
+ * snd_soc_component_get_dapm() - Returns the DAPM context associated with a
+ *  component
+ * @component: The component for which to get the DAPM context
+ */
+static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
+	struct snd_soc_component *component)
+{
+	return &component->dapm;
+}
+
+/**
+ * snd_soc_component_init_bias_level() - Initialize COMPONENT DAPM bias level
+ * @component: The COMPONENT for which to initialize the DAPM bias level
+ * @level: The DAPM level to initialize to
+ *
+ * Initializes the COMPONENT DAPM bias level. See snd_soc_dapm_init_bias_level()
+ */
+static inline void
+snd_soc_component_init_bias_level(struct snd_soc_component *component,
+				  enum snd_soc_bias_level level)
+{
+	snd_soc_dapm_init_bias_level(
+		snd_soc_component_get_dapm(component), level);
+}
+
+/**
+ * snd_soc_component_get_bias_level() - Get current COMPONENT DAPM bias level
+ * @component: The COMPONENT for which to get the DAPM bias level
+ *
+ * Returns: The current DAPM bias level of the COMPONENT.
+ */
+static inline enum snd_soc_bias_level
+snd_soc_component_get_bias_level(struct snd_soc_component *component)
+{
+	return snd_soc_dapm_get_bias_level(
+		snd_soc_component_get_dapm(component));
+}
+
+/**
+ * snd_soc_component_force_bias_level() - Set the COMPONENT DAPM bias level
+ * @component: The COMPONENT for which to set the level
+ * @level: The level to set to
+ *
+ * Forces the COMPONENT bias level to a specific state. See
+ * snd_soc_dapm_force_bias_level().
+ */
+static inline int
+snd_soc_component_force_bias_level(struct snd_soc_component *component,
+				   enum snd_soc_bias_level level)
+{
+	return snd_soc_dapm_force_bias_level(
+		snd_soc_component_get_dapm(component),
+		level);
+}
+
+/**
+ * snd_soc_dapm_kcontrol_component() - Returns the component associated to a
+ * kcontrol
+ * @kcontrol: The kcontrol
+ *
+ * This function must only be used on DAPM contexts that are known to be part of
+ * a COMPONENT (e.g. in a COMPONENT driver). Otherwise the behavior is undefined
+ */
+static inline struct snd_soc_component *snd_soc_dapm_kcontrol_component(
+	struct snd_kcontrol *kcontrol)
+{
+	return snd_soc_dapm_to_component(snd_soc_dapm_kcontrol_dapm(kcontrol));
+}
+
+/**
+ * snd_soc_component_cache_sync() - Sync the register cache with the hardware
+ * @component: COMPONENT to sync
+ *
+ * Note: This function will call regcache_sync()
+ */
+static inline int snd_soc_component_cache_sync(
+	struct snd_soc_component *component)
+{
+	return regcache_sync(component->regmap);
+}
+
+/* component IO */
+int snd_soc_component_read(struct snd_soc_component *component,
+			   unsigned int reg, unsigned int *val);
+unsigned int snd_soc_component_read32(struct snd_soc_component *component,
+				      unsigned int reg);
+int snd_soc_component_write(struct snd_soc_component *component,
+			    unsigned int reg, unsigned int val);
+int snd_soc_component_update_bits(struct snd_soc_component *component,
+				  unsigned int reg, unsigned int mask,
+				  unsigned int val);
+int snd_soc_component_update_bits_async(struct snd_soc_component *component,
+					unsigned int reg, unsigned int mask,
+					unsigned int val);
+void snd_soc_component_async_complete(struct snd_soc_component *component);
+int snd_soc_component_test_bits(struct snd_soc_component *component,
+				unsigned int reg, unsigned int mask,
+				unsigned int value);
+
+/* component wide operations */
+int snd_soc_component_set_sysclk(struct snd_soc_component *component,
+				 int clk_id, int source,
+				 unsigned int freq, int dir);
+int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
+			      int source, unsigned int freq_in,
+			      unsigned int freq_out);
+int snd_soc_component_set_jack(struct snd_soc_component *component,
+			       struct snd_soc_jack *jack, void *data);
+
+#ifdef CONFIG_REGMAP
+void snd_soc_component_init_regmap(struct snd_soc_component *component,
+				   struct regmap *regmap);
+void snd_soc_component_exit_regmap(struct snd_soc_component *component);
+#endif
+
+static inline void snd_soc_component_set_drvdata(struct snd_soc_component *c,
+						 void *data)
+{
+	dev_set_drvdata(c->dev, data);
+}
+
+static inline void *snd_soc_component_get_drvdata(struct snd_soc_component *c)
+{
+	return dev_get_drvdata(c->dev);
+}
+
+static inline bool snd_soc_component_is_active(
+	struct snd_soc_component *component)
+{
+	return component->active != 0;
+}
+
+/* component pin */
+int snd_soc_component_enable_pin(struct snd_soc_component *component,
+				 const char *pin);
+int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component,
+					  const char *pin);
+int snd_soc_component_disable_pin(struct snd_soc_component *component,
+				  const char *pin);
+int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component,
+					   const char *pin);
+int snd_soc_component_nc_pin(struct snd_soc_component *component,
+			     const char *pin);
+int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component,
+				      const char *pin);
+int snd_soc_component_get_pin_status(struct snd_soc_component *component,
+				     const char *pin);
+int snd_soc_component_force_enable_pin(struct snd_soc_component *component,
+				       const char *pin);
+int snd_soc_component_force_enable_pin_unlocked(
+	struct snd_soc_component *component,
+	const char *pin);
+
+#endif /* __SOC_COMPONENT_H */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index f0797792dd8d..6ac6481b4882 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -362,21 +362,6 @@
 #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
 	const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
 
-/*
- * Component probe and remove ordering levels for components with runtime
- * dependencies.
- */
-#define SND_SOC_COMP_ORDER_FIRST		-2
-#define SND_SOC_COMP_ORDER_EARLY		-1
-#define SND_SOC_COMP_ORDER_NORMAL		0
-#define SND_SOC_COMP_ORDER_LATE		1
-#define SND_SOC_COMP_ORDER_LAST		2
-
-#define for_each_comp_order(order)		\
-	for (order  = SND_SOC_COMP_ORDER_FIRST;	\
-	     order <= SND_SOC_COMP_ORDER_LAST;	\
-	     order++)
-
 /*
  * Bias levels
  *
@@ -747,132 +732,6 @@ struct snd_soc_compr_ops {
 	int (*trigger)(struct snd_compr_stream *);
 };
 
-/* component interface */
-struct snd_soc_component_driver {
-	const char *name;
-
-	/* Default control and setup, added after probe() is run */
-	const struct snd_kcontrol_new *controls;
-	unsigned int num_controls;
-	const struct snd_soc_dapm_widget *dapm_widgets;
-	unsigned int num_dapm_widgets;
-	const struct snd_soc_dapm_route *dapm_routes;
-	unsigned int num_dapm_routes;
-
-	int (*probe)(struct snd_soc_component *);
-	void (*remove)(struct snd_soc_component *);
-	int (*suspend)(struct snd_soc_component *);
-	int (*resume)(struct snd_soc_component *);
-
-	unsigned int (*read)(struct snd_soc_component *, unsigned int);
-	int (*write)(struct snd_soc_component *, unsigned int, unsigned int);
-
-	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_soc_pcm_runtime *);
-	void (*pcm_free)(struct snd_pcm *);
-
-	/* component wide operations */
-	int (*set_sysclk)(struct snd_soc_component *component,
-			  int clk_id, int source, unsigned int freq, int dir);
-	int (*set_pll)(struct snd_soc_component *component, int pll_id,
-		       int source, unsigned int freq_in, unsigned int freq_out);
-	int (*set_jack)(struct snd_soc_component *component,
-			struct snd_soc_jack *jack,  void *data);
-
-	/* DT */
-	int (*of_xlate_dai_name)(struct snd_soc_component *component,
-				 struct of_phandle_args *args,
-				 const char **dai_name);
-	int (*of_xlate_dai_id)(struct snd_soc_component *comment,
-			       struct device_node *endpoint);
-	void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type,
-		int subseq);
-	int (*stream_event)(struct snd_soc_component *, int event);
-	int (*set_bias_level)(struct snd_soc_component *component,
-			      enum snd_soc_bias_level level);
-
-	const struct snd_pcm_ops *ops;
-	const struct snd_compr_ops *compr_ops;
-
-	/* probe ordering - for components with runtime dependencies */
-	int probe_order;
-	int remove_order;
-
-	/*
-	 * signal if the module handling the component should not be removed
-	 * if a pcm is open. Setting this would prevent the module
-	 * refcount being incremented in probe() but allow it be incremented
-	 * when a pcm is opened and decremented when it is closed.
-	 */
-	unsigned int module_get_upon_open:1;
-
-	/* bits */
-	unsigned int idle_bias_on:1;
-	unsigned int suspend_bias_off:1;
-	unsigned int use_pmdown_time:1; /* care pmdown_time at stop */
-	unsigned int endianness:1;
-	unsigned int non_legacy_dai_naming:1;
-
-	/* this component uses topology and ignore machine driver FEs */
-	const char *ignore_machine;
-	const char *topology_name_prefix;
-	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
-				  struct snd_pcm_hw_params *params);
-	bool use_dai_pcm_id;	/* use the DAI link PCM ID as PCM device number */
-	int be_pcm_base;	/* base device ID for all BE PCMs */
-};
-
-struct snd_soc_component {
-	const char *name;
-	int id;
-	const char *name_prefix;
-	struct device *dev;
-	struct snd_soc_card *card;
-
-	unsigned int active;
-
-	unsigned int suspended:1; /* is in suspend PM state */
-
-	struct list_head list;
-	struct list_head card_aux_list; /* for auxiliary bound components */
-	struct list_head card_list;
-
-	const struct snd_soc_component_driver *driver;
-
-	struct list_head dai_list;
-	int num_dai;
-
-	struct regmap *regmap;
-	int val_bytes;
-
-	struct mutex io_mutex;
-
-	/* attached dynamic objects */
-	struct list_head dobj_list;
-
-	/*
-	* DO NOT use any of the fields below in drivers, they are temporary and
-	* are going to be removed again soon. If you use them in driver code the
-	* driver will be marked as BROKEN when these fields are removed.
-	*/
-
-	/* Don't use these, use snd_soc_component_get_dapm() */
-	struct snd_soc_dapm_context dapm;
-
-	/* machine specific init */
-	int (*init)(struct snd_soc_component *component);
-
-#ifdef CONFIG_DEBUG_FS
-	struct dentry *debugfs_root;
-	const char *debugfs_prefix;
-#endif
-};
-
-#define for_each_component_dais(component, dai)\
-	list_for_each_entry(dai, &(component)->dai_list, list)
-#define for_each_component_dais_safe(component, dai, _dai)\
-	list_for_each_entry_safe(dai, _dai, &(component)->dai_list, list)
-
 struct snd_soc_rtdcom_list {
 	struct snd_soc_component *component;
 	struct list_head list; /* rtd::component_list */
@@ -1337,134 +1196,6 @@ struct soc_enum {
 	struct snd_soc_dobj dobj;
 };
 
-/**
- * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is
- *  embedded in
- * @dapm: The DAPM context to cast to the component
- *
- * This function must only be used on DAPM contexts that are known to be part of
- * a component (e.g. in a component driver). Otherwise the behavior is
- * undefined.
- */
-static inline struct snd_soc_component *snd_soc_dapm_to_component(
-	struct snd_soc_dapm_context *dapm)
-{
-	return container_of(dapm, struct snd_soc_component, dapm);
-}
-
-/**
- * snd_soc_component_get_dapm() - Returns the DAPM context associated with a
- *  component
- * @component: The component for which to get the DAPM context
- */
-static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
-	struct snd_soc_component *component)
-{
-	return &component->dapm;
-}
-
-/**
- * snd_soc_component_init_bias_level() - Initialize COMPONENT DAPM bias level
- * @component: The COMPONENT for which to initialize the DAPM bias level
- * @level: The DAPM level to initialize to
- *
- * Initializes the COMPONENT DAPM bias level. See snd_soc_dapm_init_bias_level().
- */
-static inline void
-snd_soc_component_init_bias_level(struct snd_soc_component *component,
-				  enum snd_soc_bias_level level)
-{
-	snd_soc_dapm_init_bias_level(
-		snd_soc_component_get_dapm(component), level);
-}
-
-/**
- * snd_soc_component_get_bias_level() - Get current COMPONENT DAPM bias level
- * @component: The COMPONENT for which to get the DAPM bias level
- *
- * Returns: The current DAPM bias level of the COMPONENT.
- */
-static inline enum snd_soc_bias_level
-snd_soc_component_get_bias_level(struct snd_soc_component *component)
-{
-	return snd_soc_dapm_get_bias_level(
-		snd_soc_component_get_dapm(component));
-}
-
-/**
- * snd_soc_component_force_bias_level() - Set the COMPONENT DAPM bias level
- * @component: The COMPONENT for which to set the level
- * @level: The level to set to
- *
- * Forces the COMPONENT bias level to a specific state. See
- * snd_soc_dapm_force_bias_level().
- */
-static inline int
-snd_soc_component_force_bias_level(struct snd_soc_component *component,
-				   enum snd_soc_bias_level level)
-{
-	return snd_soc_dapm_force_bias_level(
-		snd_soc_component_get_dapm(component),
-		level);
-}
-
-/**
- * snd_soc_dapm_kcontrol_component() - Returns the component associated to a kcontrol
- * @kcontrol: The kcontrol
- *
- * This function must only be used on DAPM contexts that are known to be part of
- * a COMPONENT (e.g. in a COMPONENT driver). Otherwise the behavior is undefined.
- */
-static inline struct snd_soc_component *snd_soc_dapm_kcontrol_component(
-	struct snd_kcontrol *kcontrol)
-{
-	return snd_soc_dapm_to_component(snd_soc_dapm_kcontrol_dapm(kcontrol));
-}
-
-/**
- * snd_soc_component_cache_sync() - Sync the register cache with the hardware
- * @component: COMPONENT to sync
- *
- * Note: This function will call regcache_sync()
- */
-static inline int snd_soc_component_cache_sync(
-	struct snd_soc_component *component)
-{
-	return regcache_sync(component->regmap);
-}
-
-/* component IO */
-int snd_soc_component_read(struct snd_soc_component *component,
-	unsigned int reg, unsigned int *val);
-unsigned int snd_soc_component_read32(struct snd_soc_component *component,
-				      unsigned int reg);
-int snd_soc_component_write(struct snd_soc_component *component,
-	unsigned int reg, unsigned int val);
-int snd_soc_component_update_bits(struct snd_soc_component *component,
-	unsigned int reg, unsigned int mask, unsigned int val);
-int snd_soc_component_update_bits_async(struct snd_soc_component *component,
-	unsigned int reg, unsigned int mask, unsigned int val);
-void snd_soc_component_async_complete(struct snd_soc_component *component);
-int snd_soc_component_test_bits(struct snd_soc_component *component,
-	unsigned int reg, unsigned int mask, unsigned int value);
-
-/* component wide operations */
-int snd_soc_component_set_sysclk(struct snd_soc_component *component,
-			int clk_id, int source, unsigned int freq, int dir);
-int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
-			      int source, unsigned int freq_in,
-			      unsigned int freq_out);
-int snd_soc_component_set_jack(struct snd_soc_component *component,
-			       struct snd_soc_jack *jack, void *data);
-
-#ifdef CONFIG_REGMAP
-
-void snd_soc_component_init_regmap(struct snd_soc_component *component,
-	struct regmap *regmap);
-void snd_soc_component_exit_regmap(struct snd_soc_component *component);
-
-#endif
-
 /* device driver data */
 
 static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card,
@@ -1478,17 +1209,6 @@ static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card)
 	return card->drvdata;
 }
 
-static inline void snd_soc_component_set_drvdata(struct snd_soc_component *c,
-		void *data)
-{
-	dev_set_drvdata(c->dev, data);
-}
-
-static inline void *snd_soc_component_get_drvdata(struct snd_soc_component *c)
-{
-	return dev_get_drvdata(c->dev);
-}
-
 static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
 {
 	INIT_LIST_HEAD(&card->widgets);
@@ -1535,12 +1255,6 @@ static inline unsigned int snd_soc_enum_item_to_val(struct soc_enum *e,
 	return e->values[item];
 }
 
-static inline bool snd_soc_component_is_active(
-	struct snd_soc_component *component)
-{
-	return component->active != 0;
-}
-
 /**
  * snd_soc_kcontrol_component() - Returns the component that registered the
  *  control
@@ -1676,24 +1390,6 @@ static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm)
 	mutex_unlock(&dapm->card->dapm_mutex);
 }
 
-int snd_soc_component_enable_pin(struct snd_soc_component *component,
-				 const char *pin);
-int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component,
-					  const char *pin);
-int snd_soc_component_disable_pin(struct snd_soc_component *component,
-				  const char *pin);
-int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component,
-					   const char *pin);
-int snd_soc_component_nc_pin(struct snd_soc_component *component,
-			     const char *pin);
-int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component,
-				      const char *pin);
-int snd_soc_component_get_pin_status(struct snd_soc_component *component,
-				     const char *pin);
-int snd_soc_component_force_enable_pin(struct snd_soc_component *component,
-				       const char *pin);
-int snd_soc_component_force_enable_pin_unlocked(
-					struct snd_soc_component *component,
-					const char *pin);
+#include <sound/soc-component.h>
 
 #endif
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 919c3c027c62..250a0dea9294 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,5 @@
 # SPDX-License-Identifier: GPL-2.0
-snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o
 snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o
 snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
 
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
new file mode 100644
index 000000000000..e19f78bfb919
--- /dev/null
+++ b/sound/soc/soc-component.c
@@ -0,0 +1,269 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// soc-component.c
+//
+// Copyright (C) 2019 Renesas Electronics Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+#include <sound/soc.h>
+
+/**
+ * snd_soc_component_set_sysclk - configure COMPONENT system or master clock.
+ * @component: COMPONENT
+ * @clk_id: DAI specific clock ID
+ * @source: Source for the clock
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_component_set_sysclk(struct snd_soc_component *component,
+				 int clk_id, int source, unsigned int freq,
+				 int dir)
+{
+	if (component->driver->set_sysclk)
+		return component->driver->set_sysclk(component, clk_id, source,
+						     freq, dir);
+
+	return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_set_sysclk);
+
+/*
+ * snd_soc_component_set_pll - configure component PLL.
+ * @component: COMPONENT
+ * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
+			      int source, unsigned int freq_in,
+			      unsigned int freq_out)
+{
+	if (component->driver->set_pll)
+		return component->driver->set_pll(component, pll_id, source,
+						  freq_in, freq_out);
+
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_set_pll);
+
+int snd_soc_component_enable_pin(struct snd_soc_component *component,
+				 const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_enable_pin(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_enable_pin(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin);
+
+int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component,
+					  const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_enable_pin_unlocked(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_enable_pin_unlocked(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin_unlocked);
+
+int snd_soc_component_disable_pin(struct snd_soc_component *component,
+				  const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_disable_pin(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_disable_pin(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin);
+
+int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component,
+					   const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_disable_pin_unlocked(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_disable_pin_unlocked(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin_unlocked);
+
+int snd_soc_component_nc_pin(struct snd_soc_component *component,
+			     const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_nc_pin(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_nc_pin(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin);
+
+int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component,
+				      const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_nc_pin_unlocked(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_nc_pin_unlocked(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin_unlocked);
+
+int snd_soc_component_get_pin_status(struct snd_soc_component *component,
+				     const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_get_pin_status(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_get_pin_status(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_get_pin_status);
+
+int snd_soc_component_force_enable_pin(struct snd_soc_component *component,
+				       const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_force_enable_pin(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_force_enable_pin(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin);
+
+int snd_soc_component_force_enable_pin_unlocked(
+	struct snd_soc_component *component,
+	const char *pin)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	char *full_name;
+	int ret;
+
+	if (!component->name_prefix)
+		return snd_soc_dapm_force_enable_pin_unlocked(dapm, pin);
+
+	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
+	if (!full_name)
+		return -ENOMEM;
+
+	ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, full_name);
+	kfree(full_name);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked);
+
+/**
+ * snd_soc_component_set_jack - configure component jack.
+ * @component: COMPONENTs
+ * @jack: structure to use for the jack
+ * @data: can be used if codec driver need extra data for configuring jack
+ *
+ * Configures and enables jack detection function.
+ */
+int snd_soc_component_set_jack(struct snd_soc_component *component,
+			       struct snd_soc_jack *jack, void *data)
+{
+	if (component->driver->set_jack)
+		return component->driver->set_jack(component, jack, data);
+
+	return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_component_set_jack);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ed66d2c68d10..dc3e45547da2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2345,50 +2345,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
 }
 EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls);
 
-/**
- * snd_soc_component_set_sysclk - configure COMPONENT system or master clock.
- * @component: COMPONENT
- * @clk_id: DAI specific clock ID
- * @source: Source for the clock
- * @freq: new clock frequency in Hz
- * @dir: new clock direction - input/output.
- *
- * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
- */
-int snd_soc_component_set_sysclk(struct snd_soc_component *component,
-				 int clk_id, int source, unsigned int freq,
-				 int dir)
-{
-	if (component->driver->set_sysclk)
-		return component->driver->set_sysclk(component, clk_id, source,
-						 freq, dir);
-
-	return -ENOTSUPP;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_set_sysclk);
-
-/*
- * snd_soc_component_set_pll - configure component PLL.
- * @component: COMPONENT
- * @pll_id: DAI specific PLL ID
- * @source: DAI specific source for the PLL
- * @freq_in: PLL input clock frequency in Hz
- * @freq_out: requested PLL output clock frequency in Hz
- *
- * Configures and enables PLL to generate output clock based on input clock.
- */
-int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
-			      int source, unsigned int freq_in,
-			      unsigned int freq_out)
-{
-	if (component->driver->set_pll)
-		return component->driver->set_pll(component, pll_id, source,
-						  freq_in, freq_out);
-
-	return -EINVAL;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_set_pll);
-
 static int snd_soc_bind_card(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index c7b990abdbaa..a71d2340eb05 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -23,24 +23,6 @@ struct jack_gpio_tbl {
 	struct snd_soc_jack_gpio *gpios;
 };
 
-/**
- * snd_soc_component_set_jack - configure component jack.
- * @component: COMPONENTs
- * @jack: structure to use for the jack
- * @data: can be used if codec driver need extra data for configuring jack
- *
- * Configures and enables jack detection function.
- */
-int snd_soc_component_set_jack(struct snd_soc_component *component,
-			       struct snd_soc_jack *jack, void *data)
-{
-	if (component->driver->set_jack)
-		return component->driver->set_jack(component, jack, data);
-
-	return -ENOTSUPP;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_set_jack);
-
 /**
  * snd_soc_card_jack_new - Create a new jack
  * @card:  ASoC card
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index e3b9dd634c6d..54dcece52b0c 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -52,205 +52,6 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
 }
 EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
 
-int snd_soc_component_enable_pin(struct snd_soc_component *component,
-				 const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_enable_pin(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_enable_pin(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin);
-
-int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component,
-					  const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_enable_pin_unlocked(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_enable_pin_unlocked(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin_unlocked);
-
-int snd_soc_component_disable_pin(struct snd_soc_component *component,
-				  const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_disable_pin(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_disable_pin(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin);
-
-int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component,
-					   const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_disable_pin_unlocked(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_disable_pin_unlocked(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin_unlocked);
-
-int snd_soc_component_nc_pin(struct snd_soc_component *component,
-			     const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_nc_pin(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_nc_pin(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin);
-
-int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component,
-				      const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_nc_pin_unlocked(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_nc_pin_unlocked(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin_unlocked);
-
-int snd_soc_component_get_pin_status(struct snd_soc_component *component,
-				     const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_get_pin_status(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_get_pin_status(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_get_pin_status);
-
-int snd_soc_component_force_enable_pin(struct snd_soc_component *component,
-				       const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_force_enable_pin(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_force_enable_pin(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin);
-
-int snd_soc_component_force_enable_pin_unlocked(
-					struct snd_soc_component *component,
-					const char *pin)
-{
-	struct snd_soc_dapm_context *dapm =
-		snd_soc_component_get_dapm(component);
-	char *full_name;
-	int ret;
-
-	if (!component->name_prefix)
-		return snd_soc_dapm_force_enable_pin_unlocked(dapm, pin);
-
-	full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin);
-	if (!full_name)
-		return -ENOMEM;
-
-	ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, full_name);
-	kfree(full_name);
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked);
-
 static const struct snd_pcm_hardware dummy_dma_hardware = {
 	/* Random values to keep userspace happy when checking constraints */
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
-- 
cgit v1.2.3-59-g8ed1b


From 4a81e8f30d0b422b7f10562952124d719f73b071 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:49:54 +0900
Subject: ASoC: soc-component: add snd_soc_component_get/put()

ALSA SoC is calling try_module_get()/module_put() based on
component->driver->module_get_upon_open.
To keep simple and readable code, we should create its function.
This patch adds new snd_soc_component_get/put().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h8795ro4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h | 13 +++++++++++++
 sound/soc/soc-component.c     | 18 ++++++++++++++++++
 sound/soc/soc-core.c          |  9 ++++-----
 sound/soc/soc-pcm.c           | 10 ++++------
 4 files changed, 39 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index a97d499e5d7a..a76cadf49a16 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -287,6 +287,19 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component,
 void snd_soc_component_exit_regmap(struct snd_soc_component *component);
 #endif
 
+#define snd_soc_component_module_get_when_probe(component)\
+	snd_soc_component_module_get(component, 0)
+#define snd_soc_component_module_get_when_open(component)	\
+	snd_soc_component_module_get(component, 1)
+int snd_soc_component_module_get(struct snd_soc_component *component,
+				 int upon_open);
+#define snd_soc_component_module_put_when_remove(component)	\
+	snd_soc_component_module_put(component, 0)
+#define snd_soc_component_module_put_when_close(component)	\
+	snd_soc_component_module_put(component, 1)
+void snd_soc_component_module_put(struct snd_soc_component *component,
+				  int upon_open);
+
 static inline void snd_soc_component_set_drvdata(struct snd_soc_component *c,
 						 void *data)
 {
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index e19f78bfb919..ac2d7bd5d844 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -5,6 +5,7 @@
 // Copyright (C) 2019 Renesas Electronics Corp.
 // Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
 //
+#include <linux/module.h>
 #include <sound/soc.h>
 
 /**
@@ -267,3 +268,20 @@ int snd_soc_component_set_jack(struct snd_soc_component *component,
 	return -ENOTSUPP;
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_set_jack);
+
+int snd_soc_component_module_get(struct snd_soc_component *component,
+				 int upon_open)
+{
+	if (component->driver->module_get_upon_open == !!upon_open &&
+	    !try_module_get(component->dev->driver->owner))
+		return -ENODEV;
+
+	return 0;
+}
+
+void snd_soc_component_module_put(struct snd_soc_component *component,
+				  int upon_open)
+{
+	if (component->driver->module_get_upon_open == !!upon_open)
+		module_put(component->dev->driver->owner);
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index dc3e45547da2..9ba19efcc56c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -926,8 +926,7 @@ static void soc_cleanup_component(struct snd_soc_component *component)
 	snd_soc_dapm_free(snd_soc_component_get_dapm(component));
 	soc_cleanup_component_debugfs(component);
 	component->card = NULL;
-	if (!component->driver->module_get_upon_open)
-		module_put(component->dev->driver->owner);
+	snd_soc_component_module_put_when_close(component);
 }
 
 static void soc_remove_component(struct snd_soc_component *component)
@@ -1255,9 +1254,9 @@ static int soc_probe_component(struct snd_soc_card *card,
 		return 0;
 	}
 
-	if (!component->driver->module_get_upon_open &&
-	    !try_module_get(component->dev->driver->owner))
-		return -ENODEV;
+	ret = snd_soc_component_module_get_when_probe(component);
+	if (ret < 0)
+		return ret;
 
 	component->card = card;
 	dapm->card = card;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 1e6c4e226933..5fef18507286 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -15,7 +15,6 @@
 #include <linux/delay.h>
 #include <linux/pinctrl/consumer.h>
 #include <linux/pm_runtime.h>
-#include <linux/module.h>
 #include <linux/slab.h>
 #include <linux/workqueue.h>
 #include <linux/export.h>
@@ -440,12 +439,12 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream,
 		component = rtdcom->component;
 		*last = component;
 
-		if (component->driver->module_get_upon_open &&
-		    !try_module_get(component->dev->driver->owner)) {
+		ret = snd_soc_component_module_get_when_open(component);
+		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: can't get module %s\n",
 				component->name);
-			return -ENODEV;
+			return ret;
 		}
 
 		if (!component->driver->ops ||
@@ -481,8 +480,7 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream,
 		    component->driver->ops->close)
 			component->driver->ops->close(substream);
 
-		if (component->driver->module_get_upon_open)
-			module_put(component->dev->driver->owner);
+		snd_soc_component_module_put_when_close(component);
 	}
 
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From ae2f4849286eed48a3aa79a7b73bb5bcd0c9213b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:50:01 +0900
Subject: ASoC: soc-component: add snd_soc_component_open()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_open() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ftmt5rnx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  4 ++++
 sound/soc/soc-component.c     | 10 ++++++++++
 sound/soc/soc-pcm.c           |  6 +-----
 3 files changed, 15 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index a76cadf49a16..156b1a5b6ddd 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -338,4 +338,8 @@ int snd_soc_component_force_enable_pin_unlocked(
 	struct snd_soc_component *component,
 	const char *pin);
 
+/* component driver ops */
+int snd_soc_component_open(struct snd_soc_component *component,
+			   struct snd_pcm_substream *substream);
+
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index ac2d7bd5d844..ada46f9729b2 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -285,3 +285,13 @@ void snd_soc_component_module_put(struct snd_soc_component *component,
 	if (component->driver->module_get_upon_open == !!upon_open)
 		module_put(component->dev->driver->owner);
 }
+
+int snd_soc_component_open(struct snd_soc_component *component,
+			   struct snd_pcm_substream *substream)
+{
+	if (component->driver->ops &&
+	    component->driver->ops->open)
+		return component->driver->ops->open(substream);
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 5fef18507286..caf7028cee62 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -447,11 +447,7 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream,
 			return ret;
 		}
 
-		if (!component->driver->ops ||
-		    !component->driver->ops->open)
-			continue;
-
-		ret = component->driver->ops->open(substream);
+		ret = snd_soc_component_open(component, substream);
 		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: can't open component %s: %d\n",
-- 
cgit v1.2.3-59-g8ed1b


From 3672beb8cad6beb7b8c017514aef0f4f507debcf Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:50:07 +0900
Subject: ASoC: soc-component: add snd_soc_component_close()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_close() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ef2d5rnr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 sound/soc/soc-component.c     | 10 ++++++++++
 sound/soc/soc-pcm.c           |  8 +++-----
 3 files changed, 15 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 156b1a5b6ddd..8dacac953884 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -341,5 +341,7 @@ int snd_soc_component_force_enable_pin_unlocked(
 /* component driver ops */
 int snd_soc_component_open(struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream);
+int snd_soc_component_close(struct snd_soc_component *component,
+			    struct snd_pcm_substream *substream);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index ada46f9729b2..cee66183470d 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -295,3 +295,13 @@ int snd_soc_component_open(struct snd_soc_component *component,
 
 	return 0;
 }
+
+int snd_soc_component_close(struct snd_soc_component *component,
+			    struct snd_pcm_substream *substream)
+{
+	if (component->driver->ops &&
+	    component->driver->ops->close)
+		return component->driver->ops->close(substream);
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index caf7028cee62..4848642ef159 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -465,6 +465,7 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_component *component;
+	int ret = 0;
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -472,14 +473,11 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream,
 		if (component == last)
 			break;
 
-		if (component->driver->ops &&
-		    component->driver->ops->close)
-			component->driver->ops->close(substream);
-
+		ret |= snd_soc_component_close(component, substream);
 		snd_soc_component_module_put_when_close(component);
 	}
 
-	return 0;
+	return ret;
 }
 
 /*
-- 
cgit v1.2.3-59-g8ed1b


From 6d53723380ed73a2ced648d6e92774b39e5af1bd Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:50:13 +0900
Subject: ASoC: soc-component: add snd_soc_component_prepare()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_prepare() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87d0hx5rnm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 sound/soc/soc-component.c     | 10 ++++++++++
 sound/soc/soc-pcm.c           |  6 +-----
 3 files changed, 13 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 8dacac953884..ff8233014444 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -343,5 +343,7 @@ int snd_soc_component_open(struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream);
 int snd_soc_component_close(struct snd_soc_component *component,
 			    struct snd_pcm_substream *substream);
+int snd_soc_component_prepare(struct snd_soc_component *component,
+			      struct snd_pcm_substream *substream);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index cee66183470d..733d7139d875 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -305,3 +305,13 @@ int snd_soc_component_close(struct snd_soc_component *component,
 
 	return 0;
 }
+
+int snd_soc_component_prepare(struct snd_soc_component *component,
+			      struct snd_pcm_substream *substream)
+{
+	if (component->driver->ops &&
+	    component->driver->ops->prepare)
+		return component->driver->ops->prepare(substream);
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 4848642ef159..c7fee67bc0dc 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -785,11 +785,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
-		if (!component->driver->ops ||
-		    !component->driver->ops->prepare)
-			continue;
-
-		ret = component->driver->ops->prepare(substream);
+		ret = snd_soc_component_prepare(component, substream);
 		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: platform prepare error: %d\n", ret);
-- 
cgit v1.2.3-59-g8ed1b


From 245c539a1206d74e8508a07550fb7c99d0817b8c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:50:19 +0900
Subject: ASoC: soc-component: add snd_soc_component_hw_params()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_hw_params() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87blxh5rnf.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  3 +++
 sound/soc/soc-component.c     | 11 +++++++++++
 sound/soc/soc-pcm.c           |  6 +-----
 3 files changed, 15 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index ff8233014444..778a6e7d352d 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -345,5 +345,8 @@ int snd_soc_component_close(struct snd_soc_component *component,
 			    struct snd_pcm_substream *substream);
 int snd_soc_component_prepare(struct snd_soc_component *component,
 			      struct snd_pcm_substream *substream);
+int snd_soc_component_hw_params(struct snd_soc_component *component,
+				struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 733d7139d875..7b6456370da5 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -315,3 +315,14 @@ int snd_soc_component_prepare(struct snd_soc_component *component,
 
 	return 0;
 }
+
+int snd_soc_component_hw_params(struct snd_soc_component *component,
+				struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	if (component->driver->ops &&
+	    component->driver->ops->hw_params)
+		return component->driver->ops->hw_params(substream, params);
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index c7fee67bc0dc..8be1d22dc87a 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -951,11 +951,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
-		if (!component->driver->ops ||
-		    !component->driver->ops->hw_params)
-			continue;
-
-		ret = component->driver->ops->hw_params(substream, params);
+		ret = snd_soc_component_hw_params(component, substream, params);
 		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: %s hw params failed: %d\n",
-- 
cgit v1.2.3-59-g8ed1b


From eae7136aa2083699c69de5890fd6c32c501952b5 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:50:24 +0900
Subject: ASoC: soc-component: add snd_soc_component_hw_free()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_hw_free() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a7d15rna.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 sound/soc/soc-component.c     | 10 ++++++++++
 sound/soc/soc-pcm.c           |  9 +++------
 3 files changed, 15 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 778a6e7d352d..fbcd911ac25e 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -348,5 +348,7 @@ int snd_soc_component_prepare(struct snd_soc_component *component,
 int snd_soc_component_hw_params(struct snd_soc_component *component,
 				struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params);
+int snd_soc_component_hw_free(struct snd_soc_component *component,
+			      struct snd_pcm_substream *substream);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 7b6456370da5..e2bc34efe547 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -326,3 +326,13 @@ int snd_soc_component_hw_params(struct snd_soc_component *component,
 
 	return 0;
 }
+
+int snd_soc_component_hw_free(struct snd_soc_component *component,
+			       struct snd_pcm_substream *substream)
+{
+	if (component->driver->ops &&
+	    component->driver->ops->hw_free)
+		return component->driver->ops->hw_free(substream);
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 8be1d22dc87a..8c5289904f20 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -847,6 +847,7 @@ static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_component *component;
+	int ret = 0;
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -854,14 +855,10 @@ static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream,
 		if (component == last)
 			break;
 
-		if (!component->driver->ops ||
-		    !component->driver->ops->hw_free)
-			continue;
-
-		component->driver->ops->hw_free(substream);
+		ret |= snd_soc_component_hw_free(component, substream);
 	}
 
-	return 0;
+	return ret;
 }
 
 /*
-- 
cgit v1.2.3-59-g8ed1b


From 5693d50c830272cb3c4a04d2ce4db502debd1259 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:50:29 +0900
Subject: ASoC: soc-component: add snd_soc_component_trigger()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_trigger() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878ssl5rn5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  3 +++
 sound/soc/soc-component.c     | 11 +++++++++++
 sound/soc/soc-pcm.c           |  6 +-----
 3 files changed, 15 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index fbcd911ac25e..302e27a89d47 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -350,5 +350,8 @@ int snd_soc_component_hw_params(struct snd_soc_component *component,
 				struct snd_pcm_hw_params *params);
 int snd_soc_component_hw_free(struct snd_soc_component *component,
 			      struct snd_pcm_substream *substream);
+int snd_soc_component_trigger(struct snd_soc_component *component,
+			      struct snd_pcm_substream *substream,
+			      int cmd);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index e2bc34efe547..cf0d20a877e6 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -336,3 +336,14 @@ int snd_soc_component_hw_free(struct snd_soc_component *component,
 
 	return 0;
 }
+
+int snd_soc_component_trigger(struct snd_soc_component *component,
+			      struct snd_pcm_substream *substream,
+			      int cmd)
+{
+	if (component->driver->ops &&
+	    component->driver->ops->trigger)
+		return component->driver->ops->trigger(substream, cmd);
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 8c5289904f20..cd49c2d688c3 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1065,11 +1065,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
 
-		if (!component->driver->ops ||
-		    !component->driver->ops->trigger)
-			continue;
-
-		ret = component->driver->ops->trigger(substream, cmd);
+		ret = snd_soc_component_trigger(component, substream, cmd);
 		if (ret < 0)
 			return ret;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From 66c51573b89d0a5c1089139a2f0dd029a755c37d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:50:34 +0900
Subject: ASoC: soc-component: add snd_soc_component_suspend()

Current ALSA SoC is directly using component->driver->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_suspend() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e855rn0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h | 1 +
 sound/soc/soc-component.c     | 7 +++++++
 sound/soc/soc-core.c          | 4 +---
 3 files changed, 9 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 302e27a89d47..cdb014f41fd1 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -353,5 +353,6 @@ int snd_soc_component_hw_free(struct snd_soc_component *component,
 int snd_soc_component_trigger(struct snd_soc_component *component,
 			      struct snd_pcm_substream *substream,
 			      int cmd);
+void snd_soc_component_suspend(struct snd_soc_component *component);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index cf0d20a877e6..f0e63cd991c8 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -347,3 +347,10 @@ int snd_soc_component_trigger(struct snd_soc_component *component,
 
 	return 0;
 }
+
+void snd_soc_component_suspend(struct snd_soc_component *component)
+{
+	if (component->driver->suspend)
+		component->driver->suspend(component);
+	component->suspended = 1;
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 9ba19efcc56c..855b19abc1d2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -520,9 +520,7 @@ int snd_soc_suspend(struct device *dev)
 				/* fall through */
 
 			case SND_SOC_BIAS_OFF:
-				if (component->driver->suspend)
-					component->driver->suspend(component);
-				component->suspended = 1;
+				snd_soc_component_suspend(component);
 				if (component->regmap)
 					regcache_mark_dirty(component->regmap);
 				/* deactivate pins to sleep state */
-- 
cgit v1.2.3-59-g8ed1b


From 9a840cbac77a90e8406296aaa132ebf2c84ed9e3 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:08 +0900
Subject: ASoC: soc-component: add snd_soc_component_resume()

Current ALSA SoC is directly using component->driver->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_resume() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875znp5rm2.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h | 1 +
 sound/soc/soc-component.c     | 7 +++++++
 sound/soc/soc-core.c          | 4 +---
 3 files changed, 9 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index cdb014f41fd1..1e3b70855ba7 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -354,5 +354,6 @@ int snd_soc_component_trigger(struct snd_soc_component *component,
 			      struct snd_pcm_substream *substream,
 			      int cmd);
 void snd_soc_component_suspend(struct snd_soc_component *component);
+void snd_soc_component_resume(struct snd_soc_component *component);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index f0e63cd991c8..cbae7672b72d 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -354,3 +354,10 @@ void snd_soc_component_suspend(struct snd_soc_component *component)
 		component->driver->suspend(component);
 	component->suspended = 1;
 }
+
+void snd_soc_component_resume(struct snd_soc_component *component)
+{
+	if (component->driver->resume)
+		component->driver->resume(component);
+	component->suspended = 0;
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 855b19abc1d2..6cdfe7b2fe06 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -593,9 +593,7 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	for_each_card_components(card, component) {
 		if (component->suspended) {
-			if (component->driver->resume)
-				component->driver->resume(component);
-			component->suspended = 0;
+			snd_soc_component_resume(component);
 		}
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From e40fadbcef583808c11d2e86b8ac1c652731468e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:13 +0900
Subject: ASoC: soc-component: add snd_soc_component_is_suspended()

Current ALSA SoC is directly using component->xxx,
But, it is not good for encapsulation.
This patch adds new snd_soc_component_is_suspended() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874l395rlx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h | 1 +
 sound/soc/soc-component.c     | 5 +++++
 sound/soc/soc-core.c          | 5 ++---
 3 files changed, 8 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 1e3b70855ba7..9600dc4ca6b4 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -355,5 +355,6 @@ int snd_soc_component_trigger(struct snd_soc_component *component,
 			      int cmd);
 void snd_soc_component_suspend(struct snd_soc_component *component);
 void snd_soc_component_resume(struct snd_soc_component *component);
+int snd_soc_component_is_suspended(struct snd_soc_component *component);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index cbae7672b72d..0a9ca84d7ac6 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -361,3 +361,8 @@ void snd_soc_component_resume(struct snd_soc_component *component)
 		component->driver->resume(component);
 	component->suspended = 0;
 }
+
+int snd_soc_component_is_suspended(struct snd_soc_component *component)
+{
+	return component->suspended;
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6cdfe7b2fe06..ea93edd328a2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -503,7 +503,7 @@ int snd_soc_suspend(struct device *dev)
 		 * If there are paths active then the COMPONENT will be held
 		 * with bias _ON and should not be suspended.
 		 */
-		if (!component->suspended) {
+		if (!snd_soc_component_is_suspended(component)) {
 			switch (snd_soc_dapm_get_bias_level(dapm)) {
 			case SND_SOC_BIAS_STANDBY:
 				/*
@@ -592,9 +592,8 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	for_each_card_components(card, component) {
-		if (component->suspended) {
+		if (snd_soc_component_is_suspended(component))
 			snd_soc_component_resume(component);
-		}
 	}
 
 	for_each_card_rtds(card, rtd) {
-- 
cgit v1.2.3-59-g8ed1b


From 08e837dd9e39bd3e25b1fa1a13f6ba44040e3f0d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:17 +0900
Subject: ASoC: soc-component: add snd_soc_component_probe()

Current ALSA SoC is directly using component->driver->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_probe() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736it5rlt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  1 +
 sound/soc/soc-component.c     |  8 ++++++++
 sound/soc/soc-core.c          | 12 +++++-------
 3 files changed, 14 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 9600dc4ca6b4..34e774efcf69 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -356,5 +356,6 @@ int snd_soc_component_trigger(struct snd_soc_component *component,
 void snd_soc_component_suspend(struct snd_soc_component *component);
 void snd_soc_component_resume(struct snd_soc_component *component);
 int snd_soc_component_is_suspended(struct snd_soc_component *component);
+int snd_soc_component_probe(struct snd_soc_component *component);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 0a9ca84d7ac6..b2bfc0375193 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -366,3 +366,11 @@ int snd_soc_component_is_suspended(struct snd_soc_component *component)
 {
 	return component->suspended;
 }
+
+int snd_soc_component_probe(struct snd_soc_component *component)
+{
+	if (component->driver->probe)
+		return component->driver->probe(component);
+
+	return 0;
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ea93edd328a2..6a422ddae130 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1282,13 +1282,11 @@ static int soc_probe_component(struct snd_soc_card *card,
 		}
 	}
 
-	if (component->driver->probe) {
-		ret = component->driver->probe(component);
-		if (ret < 0) {
-			dev_err(component->dev,
-				"ASoC: failed to probe component %d\n", ret);
-			goto err_probe;
-		}
+	ret = snd_soc_component_probe(component);
+	if (ret < 0) {
+		dev_err(component->dev,
+			"ASoC: failed to probe component %d\n", ret);
+		goto err_probe;
 	}
 	WARN(dapm->idle_bias_off &&
 	     dapm->bias_level != SND_SOC_BIAS_OFF,
-- 
cgit v1.2.3-59-g8ed1b


From 03b34dd7d87ce3493cb1837c9e59c3b3aac4724f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:22 +0900
Subject: ASoC: soc-component: add snd_soc_component_remove()

Current ALSA SoC is directly using component->driver->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_remove() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871ryd5rlo.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h | 1 +
 sound/soc/soc-component.c     | 6 ++++++
 sound/soc/soc-core.c          | 3 +--
 3 files changed, 8 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 34e774efcf69..b8480d947901 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -357,5 +357,6 @@ void snd_soc_component_suspend(struct snd_soc_component *component);
 void snd_soc_component_resume(struct snd_soc_component *component);
 int snd_soc_component_is_suspended(struct snd_soc_component *component);
 int snd_soc_component_probe(struct snd_soc_component *component);
+void snd_soc_component_remove(struct snd_soc_component *component);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index b2bfc0375193..eba77ea2b62d 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -374,3 +374,9 @@ int snd_soc_component_probe(struct snd_soc_component *component)
 
 	return 0;
 }
+
+void snd_soc_component_remove(struct snd_soc_component *component)
+{
+	if (component->driver->remove)
+		component->driver->remove(component);
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6a422ddae130..6a6403ddf62d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -929,8 +929,7 @@ static void soc_remove_component(struct snd_soc_component *component)
 	if (!component->card)
 		return;
 
-	if (component->driver->remove)
-		component->driver->remove(component);
+	snd_soc_component_remove(component);
 
 	soc_cleanup_component(component);
 }
-- 
cgit v1.2.3-59-g8ed1b


From 2c7b1704819435d188c7697c6815f788bf9e6200 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:26 +0900
Subject: ASoC: soc-component: add snd_soc_component_of_xlate_dai_id()

Current ALSA SoC is directly using component->driver->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_of_xlate_dai_id() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhl14d14.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h | 2 ++
 sound/soc/soc-component.c     | 9 +++++++++
 sound/soc/soc-core.c          | 5 ++---
 3 files changed, 13 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index b8480d947901..3f4acd337c4a 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -358,5 +358,7 @@ void snd_soc_component_resume(struct snd_soc_component *component);
 int snd_soc_component_is_suspended(struct snd_soc_component *component);
 int snd_soc_component_probe(struct snd_soc_component *component);
 void snd_soc_component_remove(struct snd_soc_component *component);
+int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component,
+				      struct device_node *ep);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index eba77ea2b62d..faf49992f661 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -380,3 +380,12 @@ void snd_soc_component_remove(struct snd_soc_component *component)
 	if (component->driver->remove)
 		component->driver->remove(component);
 }
+
+int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component,
+				      struct device_node *ep)
+{
+	if (component->driver->of_xlate_dai_id)
+		return component->driver->of_xlate_dai_id(component, ep);
+
+	return -ENOTSUPP;
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6a6403ddf62d..f63d09dd55f4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3334,9 +3334,8 @@ int snd_soc_get_dai_id(struct device_node *ep)
 	ret = -ENOTSUPP;
 	mutex_lock(&client_mutex);
 	component = soc_find_component(&dlc);
-	if (component &&
-	    component->driver->of_xlate_dai_id)
-		ret = component->driver->of_xlate_dai_id(component, ep);
+	if (component)
+		ret = snd_soc_component_of_xlate_dai_id(component, ep);
 	mutex_unlock(&client_mutex);
 
 	of_node_put(dlc.of_node);
-- 
cgit v1.2.3-59-g8ed1b


From a2a341752558cc67d6fe5c8ada7c16f9c3690f89 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:31 +0900
Subject: ASoC: soc-component: add snd_soc_component_of_xlate_dai_name()

Current ALSA SoC is directly using component->driver->xxx,
thus, it is deep nested, and makes code difficult to read,
and is not good for encapsulation.
This patch adds new snd_soc_component_of_xlate_dai_name() and use it

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y30l4d0z.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  3 +++
 sound/soc/soc-component.c     | 10 ++++++++++
 sound/soc/soc-core.c          |  7 ++-----
 3 files changed, 15 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 3f4acd337c4a..3ed2c39e45c2 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -360,5 +360,8 @@ int snd_soc_component_probe(struct snd_soc_component *component);
 void snd_soc_component_remove(struct snd_soc_component *component);
 int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component,
 				      struct device_node *ep);
+int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component,
+					struct of_phandle_args *args,
+					const char **dai_name);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index faf49992f661..de1bc5196f67 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -389,3 +389,13 @@ int snd_soc_component_of_xlate_dai_id(struct snd_soc_component *component,
 
 	return -ENOTSUPP;
 }
+
+int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component,
+					struct of_phandle_args *args,
+					const char **dai_name)
+{
+	if (component->driver->of_xlate_dai_name)
+		return component->driver->of_xlate_dai_name(component,
+						     args, dai_name);
+	return -ENOTSUPP;
+}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f63d09dd55f4..2f068c239f34 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3358,11 +3358,8 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
 		if (component_of_node != args->np)
 			continue;
 
-		if (pos->driver->of_xlate_dai_name) {
-			ret = pos->driver->of_xlate_dai_name(pos,
-							     args,
-							     dai_name);
-		} else {
+		ret = snd_soc_component_of_xlate_dai_name(pos, args, dai_name);
+		if (ret == -ENOTSUPP) {
 			struct snd_soc_dai *dai;
 			int id = -1;
 
-- 
cgit v1.2.3-59-g8ed1b


From 9d415fbf773f162a5c274e671741c6fa94b74287 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:35 +0900
Subject: ASoC: soc-component: move snd_soc_component_seq_notifier()

Current soc-dapm / soc-core are using a long way round to call
.seq_notifier.

	if (driver->seq_notifier)
		dapm->seq_notifier = ...;
	...
	if (dapm->seq_notifier)
		ret = dapm->seq_notifier(...);

We can directly call it via driver->seq_notifier.
One note here is that both Card and Component have dapm,
but, Card's dapm doesn't have dapm->component.
We need to check it.

This patch moves snd_soc_component_seq_notifier() to soc-component.c,
and updates parameters.
dapm->seq_notifier is no longer needed

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wog54d0v.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  3 +++
 include/sound/soc-dapm.h      |  2 --
 sound/soc/soc-component.c     |  7 +++++++
 sound/soc/soc-core.c          | 10 ----------
 sound/soc/soc-dapm.c          | 15 ++++++++-------
 5 files changed, 18 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 3ed2c39e45c2..7ac903c1e33f 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -281,6 +281,9 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
 int snd_soc_component_set_jack(struct snd_soc_component *component,
 			       struct snd_soc_jack *jack, void *data);
 
+void snd_soc_component_seq_notifier(struct snd_soc_component *component,
+				    enum snd_soc_dapm_type type, int subseq);
+
 #ifdef CONFIG_REGMAP
 void snd_soc_component_init_regmap(struct snd_soc_component *component,
 				   struct regmap *regmap);
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 6c6694160130..a03db6f8faa8 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -661,8 +661,6 @@ struct snd_soc_dapm_context {
 	unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
 	/* Go to BIAS_OFF in suspend if the DAPM context is idle */
 	unsigned int suspend_bias_off:1;
-	void (*seq_notifier)(struct snd_soc_dapm_context *,
-			     enum snd_soc_dapm_type, int);
 
 	struct device *dev; /* from parent - for debug */
 	struct snd_soc_component *component; /* parent component */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index de1bc5196f67..ca0b28b1d918 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -52,6 +52,13 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
 }
 EXPORT_SYMBOL_GPL(snd_soc_component_set_pll);
 
+void snd_soc_component_seq_notifier(struct snd_soc_component *component,
+				    enum snd_soc_dapm_type type, int subseq)
+{
+	if (component->driver->seq_notifier)
+		component->driver->seq_notifier(component, type, subseq);
+}
+
 int snd_soc_component_enable_pin(struct snd_soc_component *component,
 				 const char *pin)
 {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2f068c239f34..c618fecc3d45 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2646,14 +2646,6 @@ int snd_soc_register_dai(struct snd_soc_component *component,
 }
 EXPORT_SYMBOL_GPL(snd_soc_register_dai);
 
-static void snd_soc_component_seq_notifier(struct snd_soc_dapm_context *dapm,
-	enum snd_soc_dapm_type type, int subseq)
-{
-	struct snd_soc_component *component = dapm->component;
-
-	component->driver->seq_notifier(component, type, subseq);
-}
-
 static int snd_soc_component_stream_event(struct snd_soc_dapm_context *dapm,
 	int event)
 {
@@ -2690,8 +2682,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	dapm->bias_level = SND_SOC_BIAS_OFF;
 	dapm->idle_bias_off = !driver->idle_bias_on;
 	dapm->suspend_bias_off = driver->suspend_bias_off;
-	if (driver->seq_notifier)
-		dapm->seq_notifier = snd_soc_component_seq_notifier;
 	if (driver->stream_event)
 		dapm->stream_event = snd_soc_component_stream_event;
 	if (driver->set_bias_level)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d93c1038fab0..0b60f688b433 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1611,12 +1611,12 @@ static void dapm_seq_run(struct snd_soc_card *card,
 			if (!list_empty(&pending))
 				dapm_seq_run_coalesced(card, &pending);
 
-			if (cur_dapm && cur_dapm->seq_notifier) {
+			if (cur_dapm && cur_dapm->component) {
 				for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
 					if (sort[i] == cur_sort)
-						cur_dapm->seq_notifier(cur_dapm,
-								       i,
-								       cur_subseq);
+						snd_soc_component_seq_notifier(
+							cur_dapm->component,
+							i, cur_subseq);
 			}
 
 			if (cur_dapm && w->dapm != cur_dapm)
@@ -1674,11 +1674,12 @@ static void dapm_seq_run(struct snd_soc_card *card,
 	if (!list_empty(&pending))
 		dapm_seq_run_coalesced(card, &pending);
 
-	if (cur_dapm && cur_dapm->seq_notifier) {
+	if (cur_dapm && cur_dapm->component) {
 		for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
 			if (sort[i] == cur_sort)
-				cur_dapm->seq_notifier(cur_dapm,
-						       i, cur_subseq);
+				snd_soc_component_seq_notifier(
+					cur_dapm->component,
+					i, cur_subseq);
 	}
 
 	list_for_each_entry(d, &card->dapm_list, list) {
-- 
cgit v1.2.3-59-g8ed1b


From 8e2a990d76aced95c6f01c2d67d8835c86f0ca67 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:39 +0900
Subject: ASoC: soc-component: move snd_soc_component_stream_event()

Current soc-dapm / soc-core are using a long way round to call
.stream_event.

	if (driver->stream_event)
		dapm->stream_event = ...;
	...
	if (dapm->stream_event)
		ret = dapm->stream_event(...);

We can directly call it via driver->stream_event.
One note here is that both Card and Component have dapm,
but, Card's dapm doesn't have dapm->component.
We need to check it.

This patch moves snd_soc_component_stream_event() to soc-component.c
and updates parameters.
dapm->stream_event is no longer needed

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v9vp4d0r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 include/sound/soc-dapm.h      |  1 -
 sound/soc/soc-component.c     |  9 +++++++++
 sound/soc/soc-core.c          | 10 ----------
 sound/soc/soc-dapm.c          |  9 +++++++--
 5 files changed, 18 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 7ac903c1e33f..1f84f04e2670 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -283,6 +283,8 @@ int snd_soc_component_set_jack(struct snd_soc_component *component,
 
 void snd_soc_component_seq_notifier(struct snd_soc_component *component,
 				    enum snd_soc_dapm_type type, int subseq);
+int snd_soc_component_stream_event(struct snd_soc_component *component,
+				   int event);
 
 #ifdef CONFIG_REGMAP
 void snd_soc_component_init_regmap(struct snd_soc_component *component,
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index a03db6f8faa8..c2f14a335891 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -670,7 +670,6 @@ struct snd_soc_dapm_context {
 	enum snd_soc_bias_level target_bias_level;
 	struct list_head list;
 
-	int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
 	int (*set_bias_level)(struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index ca0b28b1d918..f33dda8023ec 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -59,6 +59,15 @@ void snd_soc_component_seq_notifier(struct snd_soc_component *component,
 		component->driver->seq_notifier(component, type, subseq);
 }
 
+int snd_soc_component_stream_event(struct snd_soc_component *component,
+				   int event)
+{
+	if (component->driver->stream_event)
+		return component->driver->stream_event(component, event);
+
+	return 0;
+}
+
 int snd_soc_component_enable_pin(struct snd_soc_component *component,
 				 const char *pin)
 {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c618fecc3d45..8cfbe5fb5921 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2646,14 +2646,6 @@ int snd_soc_register_dai(struct snd_soc_component *component,
 }
 EXPORT_SYMBOL_GPL(snd_soc_register_dai);
 
-static int snd_soc_component_stream_event(struct snd_soc_dapm_context *dapm,
-	int event)
-{
-	struct snd_soc_component *component = dapm->component;
-
-	return component->driver->stream_event(component, event);
-}
-
 static int snd_soc_component_set_bias_level(struct snd_soc_dapm_context *dapm,
 					enum snd_soc_bias_level level)
 {
@@ -2682,8 +2674,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	dapm->bias_level = SND_SOC_BIAS_OFF;
 	dapm->idle_bias_off = !driver->idle_bias_on;
 	dapm->suspend_bias_off = driver->suspend_bias_off;
-	if (driver->stream_event)
-		dapm->stream_event = snd_soc_component_stream_event;
 	if (driver->set_bias_level)
 		dapm->set_bias_level = snd_soc_component_set_bias_level;
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0b60f688b433..9288b2b43f98 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1913,6 +1913,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 	LIST_HEAD(down_list);
 	ASYNC_DOMAIN_EXCLUSIVE(async_domain);
 	enum snd_soc_bias_level bias;
+	int ret;
 
 	lockdep_assert_held(&card->dapm_mutex);
 
@@ -2029,8 +2030,12 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 
 	/* do we need to notify any clients that DAPM event is complete */
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d->stream_event)
-			d->stream_event(d, event);
+		if (!d->component)
+			continue;
+
+		ret = snd_soc_component_stream_event(d->component, event);
+		if (ret < 0)
+			return ret;
 	}
 
 	pop_dbg(card->dev, card->pop_time,
-- 
cgit v1.2.3-59-g8ed1b


From 7951b14611851bdae18e9bca18015b1d84731d0d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:43 +0900
Subject: ASoC: soc-component: move snd_soc_component_set_bias_level()

Current soc-dapm / soc-core are using a long way round to call
.set_bias_level.

	if (driver->set_bias_level)
		dapm->set_bias_level = ...;
	...
	if (dapm->set_bias_level)
		ret = dapm->set_bias_level(...);

We can directly call it via driver->set_bias_level.
One note here is that both Card and Component have dapm,
but, Card's dapm doesn't have dapm->component.
We need to check it.

This patch moves snd_soc_component_set_bias_level() to soc-component.c
and updates parameters.
dapm->set_bias_level is no longer needed

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tvb94d0n.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 include/sound/soc-dapm.h      |  3 ---
 sound/soc/soc-component.c     |  9 +++++++++
 sound/soc/soc-core.c          | 10 ----------
 sound/soc/soc-dapm.c          |  4 ++--
 5 files changed, 13 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 1f84f04e2670..2aaf12bbbed0 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -285,6 +285,8 @@ void snd_soc_component_seq_notifier(struct snd_soc_component *component,
 				    enum snd_soc_dapm_type type, int subseq);
 int snd_soc_component_stream_event(struct snd_soc_component *component,
 				   int event);
+int snd_soc_component_set_bias_level(struct snd_soc_component *component,
+				     enum snd_soc_bias_level level);
 
 #ifdef CONFIG_REGMAP
 void snd_soc_component_init_regmap(struct snd_soc_component *component,
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c2f14a335891..2aa73d6dd7be 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -670,9 +670,6 @@ struct snd_soc_dapm_context {
 	enum snd_soc_bias_level target_bias_level;
 	struct list_head list;
 
-	int (*set_bias_level)(struct snd_soc_dapm_context *dapm,
-			      enum snd_soc_bias_level level);
-
 	struct snd_soc_dapm_wcache path_sink_cache;
 	struct snd_soc_dapm_wcache path_source_cache;
 
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index f33dda8023ec..cb63df6e46eb 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -68,6 +68,15 @@ int snd_soc_component_stream_event(struct snd_soc_component *component,
 	return 0;
 }
 
+int snd_soc_component_set_bias_level(struct snd_soc_component *component,
+				     enum snd_soc_bias_level level)
+{
+	if (component->driver->set_bias_level)
+		return component->driver->set_bias_level(component, level);
+
+	return 0;
+}
+
 int snd_soc_component_enable_pin(struct snd_soc_component *component,
 				 const char *pin)
 {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 8cfbe5fb5921..0f75dac4bb26 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2646,14 +2646,6 @@ int snd_soc_register_dai(struct snd_soc_component *component,
 }
 EXPORT_SYMBOL_GPL(snd_soc_register_dai);
 
-static int snd_soc_component_set_bias_level(struct snd_soc_dapm_context *dapm,
-					enum snd_soc_bias_level level)
-{
-	struct snd_soc_component *component = dapm->component;
-
-	return component->driver->set_bias_level(component, level);
-}
-
 static int snd_soc_component_initialize(struct snd_soc_component *component,
 	const struct snd_soc_component_driver *driver, struct device *dev)
 {
@@ -2674,8 +2666,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	dapm->bias_level = SND_SOC_BIAS_OFF;
 	dapm->idle_bias_off = !driver->idle_bias_on;
 	dapm->suspend_bias_off = driver->suspend_bias_off;
-	if (driver->set_bias_level)
-		dapm->set_bias_level = snd_soc_component_set_bias_level;
 
 	INIT_LIST_HEAD(&component->dai_list);
 	mutex_init(&component->io_mutex);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 9288b2b43f98..d09bdca63c62 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -684,8 +684,8 @@ int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm,
 {
 	int ret = 0;
 
-	if (dapm->set_bias_level)
-		ret = dapm->set_bias_level(dapm, level);
+	if (dapm->component)
+		ret = snd_soc_component_set_bias_level(dapm->component, level);
 
 	if (ret == 0)
 		dapm->bias_level = level;
-- 
cgit v1.2.3-59-g8ed1b


From 0035e2565b93e0902a06320ba1716bc1ddd753b3 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:47 +0900
Subject: ASoC: soc-component: add snd_soc_pcm_component_pointer()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, the code nested deeply, and it makes code difficult to read,
and is not good for encapsulation.

We want to implement component related function at soc-component.c,
but, some of them need to care whole snd_soc_pcm_runtime (= rtd)
connected component.

Let's call component related function which need to care with
for_each_rtdcom() loop as snd_soc_pcm_component_xxx().
This patch adds new snd_soc_pcm_component_pointer() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgqt4d0j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 sound/soc/soc-component.c     | 18 ++++++++++++++++++
 sound/soc/soc-pcm.c           | 13 +------------
 3 files changed, 21 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 2aaf12bbbed0..38b4be1d99f5 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -371,4 +371,6 @@ int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component,
 					struct of_phandle_args *args,
 					const char **dai_name);
 
+int snd_soc_pcm_component_pointer(struct snd_pcm_substream *substream);
+
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index cb63df6e46eb..e2053c8bf1f0 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -424,3 +424,21 @@ int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component,
 						     args, dai_name);
 	return -ENOTSUPP;
 }
+
+int snd_soc_pcm_component_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_component *component;
+	struct snd_soc_rtdcom_list *rtdcom;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		/* FIXME: use 1st pointer */
+		if (component->driver->ops &&
+		    component->driver->ops->pointer)
+			return component->driver->ops->pointer(substream);
+	}
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index cd49c2d688c3..020e1d275076 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1111,8 +1111,6 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
 static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component;
-	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	struct snd_pcm_runtime *runtime = substream->runtime;
@@ -1124,17 +1122,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	/* clearing the previous total delay */
 	runtime->delay = 0;
 
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
+	offset = snd_soc_pcm_component_pointer(substream);
 
-		if (!component->driver->ops ||
-		    !component->driver->ops->pointer)
-			continue;
-
-		/* FIXME: use 1st pointer */
-		offset = component->driver->ops->pointer(substream);
-		break;
-	}
 	/* base delay if assigned in pointer callback */
 	delay = runtime->delay;
 
-- 
cgit v1.2.3-59-g8ed1b


From 96a47908d8769479f5217bf3f432ccdc06a29747 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:51 +0900
Subject: ASoC: soc-component: add snd_soc_pcm_component_ioctrl()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, the code nested deeply, and it makes code difficult to read,
and is not good for encapsulation.

We want to implement component related function at soc-component.c,
but, some of them need to care whole snd_soc_pcm_runtime (= rtd)
connected component.

Let's call component related function which need to care with
for_each_rtdcom() loop as snd_soc_pcm_component_xxx().
This patch adds new snd_soc_pcm_component_ioctrl() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r26d4d0f.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 sound/soc/soc-component.c     | 20 ++++++++++++++++++++
 sound/soc/soc-pcm.c           | 25 ++-----------------------
 3 files changed, 24 insertions(+), 23 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 38b4be1d99f5..5db4e5d028d0 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -372,5 +372,7 @@ int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component,
 					const char **dai_name);
 
 int snd_soc_pcm_component_pointer(struct snd_pcm_substream *substream);
+int snd_soc_pcm_component_ioctl(struct snd_pcm_substream *substream,
+				unsigned int cmd, void *arg);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index e2053c8bf1f0..a6c0857a9e90 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -442,3 +442,23 @@ int snd_soc_pcm_component_pointer(struct snd_pcm_substream *substream)
 
 	return 0;
 }
+
+int snd_soc_pcm_component_ioctl(struct snd_pcm_substream *substream,
+				unsigned int cmd, void *arg)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_component *component;
+	struct snd_soc_rtdcom_list *rtdcom;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		/* FIXME: use 1st ioctl */
+		if (component->driver->ops &&
+		    component->driver->ops->ioctl)
+			return component->driver->ops->ioctl(substream,
+							     cmd, arg);
+	}
+
+	return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 020e1d275076..12377b8d41c2 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2453,27 +2453,6 @@ out:
 	return ret;
 }
 
-static int soc_pcm_ioctl(struct snd_pcm_substream *substream,
-		     unsigned int cmd, void *arg)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component;
-	struct snd_soc_rtdcom_list *rtdcom;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->ioctl)
-			continue;
-
-		/* FIXME: use 1st ioctl */
-		return component->driver->ops->ioctl(substream, cmd, arg);
-	}
-
-	return snd_pcm_lib_ioctl(substream, cmd, arg);
-}
-
 static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_pcm_substream *substream =
@@ -3013,7 +2992,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		rtd->ops.hw_free	= dpcm_fe_dai_hw_free;
 		rtd->ops.close		= dpcm_fe_dai_close;
 		rtd->ops.pointer	= soc_pcm_pointer;
-		rtd->ops.ioctl		= soc_pcm_ioctl;
+		rtd->ops.ioctl		= snd_soc_pcm_component_ioctl;
 	} else {
 		rtd->ops.open		= soc_pcm_open;
 		rtd->ops.hw_params	= soc_pcm_hw_params;
@@ -3022,7 +3001,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		rtd->ops.hw_free	= soc_pcm_hw_free;
 		rtd->ops.close		= soc_pcm_close;
 		rtd->ops.pointer	= soc_pcm_pointer;
-		rtd->ops.ioctl		= soc_pcm_ioctl;
+		rtd->ops.ioctl		= snd_soc_pcm_component_ioctl;
 	}
 
 	for_each_rtdcom(rtd, rtdcom) {
-- 
cgit v1.2.3-59-g8ed1b


From 82d81f5cced36e480b581ae51c595e2deb9f2d56 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:51:56 +0900
Subject: ASoC: soc-component: add snd_soc_pcm_component_copy_user()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, the code nested deeply, and it makes code difficult to read,
and is not good for encapsulation.

We want to implement component related function at soc-component.c,
but, some of them need to care whole snd_soc_pcm_runtime (= rtd)
connected component.

Let's call component related function which need to care with
for_each_rtdcom() loop as snd_soc_pcm_component_xxx().
This patch adds new snd_soc_pcm_component_copy_user() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pnlx4d0a.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  3 +++
 sound/soc/soc-component.c     | 21 +++++++++++++++++++++
 sound/soc/soc-pcm.c           | 25 +------------------------
 3 files changed, 25 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 5db4e5d028d0..6b95d2467053 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -374,5 +374,8 @@ int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component,
 int snd_soc_pcm_component_pointer(struct snd_pcm_substream *substream);
 int snd_soc_pcm_component_ioctl(struct snd_pcm_substream *substream,
 				unsigned int cmd, void *arg);
+int snd_soc_pcm_component_copy_user(struct snd_pcm_substream *substream,
+				    int channel, unsigned long pos,
+				    void __user *buf, unsigned long bytes);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index a6c0857a9e90..20897dce1bec 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -462,3 +462,24 @@ int snd_soc_pcm_component_ioctl(struct snd_pcm_substream *substream,
 
 	return snd_pcm_lib_ioctl(substream, cmd, arg);
 }
+
+int snd_soc_pcm_component_copy_user(struct snd_pcm_substream *substream,
+				    int channel, unsigned long pos,
+				    void __user *buf, unsigned long bytes)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		/* FIXME. it returns 1st copy now */
+		if (component->driver->ops &&
+		    component->driver->ops->copy_user)
+			return component->driver->ops->copy_user(
+				substream, channel, pos, buf, bytes);
+	}
+
+	return -EINVAL;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 12377b8d41c2..b0e6ce89b012 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2818,29 +2818,6 @@ static void soc_pcm_private_free(struct snd_pcm *pcm)
 	}
 }
 
-static int soc_rtdcom_copy_user(struct snd_pcm_substream *substream, int channel,
-				unsigned long pos, void __user *buf,
-				unsigned long bytes)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->copy_user)
-			continue;
-
-		/* FIXME. it returns 1st copy now */
-		return component->driver->ops->copy_user(substream, channel,
-							 pos, buf, bytes);
-	}
-
-	return -EINVAL;
-}
-
 static struct page *soc_rtdcom_page(struct snd_pcm_substream *substream,
 				    unsigned long offset)
 {
@@ -3011,7 +2988,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			continue;
 
 		if (ops->copy_user)
-			rtd->ops.copy_user	= soc_rtdcom_copy_user;
+			rtd->ops.copy_user	= snd_soc_pcm_component_copy_user;
 		if (ops->page)
 			rtd->ops.page		= soc_rtdcom_page;
 		if (ops->mmap)
-- 
cgit v1.2.3-59-g8ed1b


From 9c712e4f57229081e837d593fc1e4183b068a41c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:52:00 +0900
Subject: ASoC: soc-component: add snd_soc_pcm_component_page()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, the code nested deeply, and it makes code difficult to read,
and is not good for encapsulation.

We want to implement component related function at soc-component.c,
but, some of them need to care whole snd_soc_pcm_runtime (= rtd)
connected component.

Let's call component related function which need to care with
for_each_rtdcom() loop as snd_soc_pcm_component_xxx().
This patch adds new snd_soc_pcm_component_page() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o91h4d06.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 sound/soc/soc-component.c     | 23 +++++++++++++++++++++++
 sound/soc/soc-pcm.c           | 26 +-------------------------
 3 files changed, 26 insertions(+), 25 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 6b95d2467053..4cab257962a6 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -377,5 +377,7 @@ int snd_soc_pcm_component_ioctl(struct snd_pcm_substream *substream,
 int snd_soc_pcm_component_copy_user(struct snd_pcm_substream *substream,
 				    int channel, unsigned long pos,
 				    void __user *buf, unsigned long bytes);
+struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
+					unsigned long offset);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 20897dce1bec..d503bc9b0850 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -483,3 +483,26 @@ int snd_soc_pcm_component_copy_user(struct snd_pcm_substream *substream,
 
 	return -EINVAL;
 }
+
+struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
+					unsigned long offset)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+	struct page *page;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		/* FIXME. it returns 1st page now */
+		if (component->driver->ops &&
+		    component->driver->ops->page) {
+			page = component->driver->ops->page(substream, offset);
+			if (page)
+				return page;
+		}
+	}
+
+	return NULL;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b0e6ce89b012..fe34f2e5d75e 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2818,30 +2818,6 @@ static void soc_pcm_private_free(struct snd_pcm *pcm)
 	}
 }
 
-static struct page *soc_rtdcom_page(struct snd_pcm_substream *substream,
-				    unsigned long offset)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
-	struct page *page;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->page)
-			continue;
-
-		/* FIXME. it returns 1st page now */
-		page = component->driver->ops->page(substream, offset);
-		if (page)
-			return page;
-	}
-
-	return NULL;
-}
-
 static int soc_rtdcom_mmap(struct snd_pcm_substream *substream,
 			   struct vm_area_struct *vma)
 {
@@ -2990,7 +2966,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		if (ops->copy_user)
 			rtd->ops.copy_user	= snd_soc_pcm_component_copy_user;
 		if (ops->page)
-			rtd->ops.page		= soc_rtdcom_page;
+			rtd->ops.page		= snd_soc_pcm_component_page;
 		if (ops->mmap)
 			rtd->ops.mmap		= soc_rtdcom_mmap;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From 205875e1a12ef9c61e939db9ded90fe3f6352e75 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:52:04 +0900
Subject: ASoC: soc-component: add snd_soc_pcm_component_mmap()

Current ALSA SoC is directly using component->driver->ops->xxx,
thus, the code nested deeply, and it makes code difficult to read,
and is not good for encapsulation.

We want to implement component related function at soc-component.c,
but, some of them need to care whole snd_soc_pcm_runtime (= rtd)
connected component.

Let's call component related function which need to care with
for_each_rtdcom() loop as snd_soc_pcm_component_xxx().
This patch adds new snd_soc_pcm_component_mmap() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87muh14d02.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  2 ++
 sound/soc/soc-component.c     | 19 +++++++++++++++++++
 sound/soc/soc-pcm.c           | 23 +----------------------
 3 files changed, 22 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 4cab257962a6..dd1ea5d71998 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -379,5 +379,7 @@ int snd_soc_pcm_component_copy_user(struct snd_pcm_substream *substream,
 				    void __user *buf, unsigned long bytes);
 struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
 					unsigned long offset);
+int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
+			       struct vm_area_struct *vma);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index d503bc9b0850..2aff1b087522 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -506,3 +506,22 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
 
 	return NULL;
 }
+
+int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
+			       struct vm_area_struct *vma)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		/* FIXME. it returns 1st mmap now */
+		if (component->driver->ops &&
+		    component->driver->ops->mmap)
+			return component->driver->ops->mmap(substream, vma);
+	}
+
+	return -EINVAL;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fe34f2e5d75e..7bbee0d71942 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2818,27 +2818,6 @@ static void soc_pcm_private_free(struct snd_pcm *pcm)
 	}
 }
 
-static int soc_rtdcom_mmap(struct snd_pcm_substream *substream,
-			   struct vm_area_struct *vma)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->ops ||
-		    !component->driver->ops->mmap)
-			continue;
-
-		/* FIXME. it returns 1st mmap now */
-		return component->driver->ops->mmap(substream, vma);
-	}
-
-	return -EINVAL;
-}
-
 /* create a new pcm */
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 {
@@ -2968,7 +2947,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		if (ops->page)
 			rtd->ops.page		= snd_soc_pcm_component_page;
 		if (ops->mmap)
-			rtd->ops.mmap		= soc_rtdcom_mmap;
+			rtd->ops.mmap		= snd_soc_pcm_component_mmap;
 	}
 
 	if (playback)
-- 
cgit v1.2.3-59-g8ed1b


From 7484291e9b7564af65b2581dcdebeeaf98bc86d0 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:52:08 +0900
Subject: ASoC: soc-component: add snd_soc_pcm_component_pcm_new()

Current ALSA SoC is directly using component->driver->xxx,
thus, the code nested deeply, and it makes code difficult to read,
and is not good for encapsulation.

We want to implement component related function at soc-component.c,
but, some of them need to care whole snd_soc_pcm_runtime (= rtd)
connected component.

Let's call component related function which need to care with
for_each_rtdcom() loop as snd_soc_pcm_component_xxx().
This patch adds new snd_soc_pcm_component_pcm() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lfwl4czy.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  1 +
 sound/soc/soc-component.c     | 20 ++++++++++++++++++++
 sound/soc/soc-pcm.c           | 18 ++++--------------
 3 files changed, 25 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index dd1ea5d71998..d3048ad06582 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -381,5 +381,6 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
 					unsigned long offset);
 int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
 			       struct vm_area_struct *vma);
+int snd_soc_pcm_component_new(struct snd_pcm *pcm);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 2aff1b087522..ff13d901bbab 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -525,3 +525,23 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
 
 	return -EINVAL;
 }
+
+int snd_soc_pcm_component_new(struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+	int ret;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (component->driver->pcm_new) {
+			ret = component->driver->pcm_new(rtd);
+			if (ret < 0)
+				return ret;
+		}
+	}
+
+	return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 7bbee0d71942..955c49fd3bda 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2823,7 +2823,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 {
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_component *component;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_pcm *pcm;
 	char new_name[64];
@@ -2956,19 +2955,10 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	if (capture)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops);
 
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (!component->driver->pcm_new)
-			continue;
-
-		ret = component->driver->pcm_new(rtd);
-		if (ret < 0) {
-			dev_err(component->dev,
-				"ASoC: pcm constructor failed: %d\n",
-				ret);
-			return ret;
-		}
+	ret = snd_soc_pcm_component_new(pcm);
+	if (ret < 0) {
+		dev_err(rtd->dev, "ASoC: pcm constructor failed: %d\n", ret);
+		return ret;
 	}
 
 	pcm->private_free = soc_pcm_private_free;
-- 
cgit v1.2.3-59-g8ed1b


From 79776da0989733a5bac0a1e635e3a284c3f5c745 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 26 Jul 2019 13:52:12 +0900
Subject: ASoC: soc-component: add snd_soc_pcm_component_pcm_free()

Current ALSA SoC is directly using component->driver->xxx,
thus, the code nested deeply, and it makes code difficult to read,
and is not good for encapsulation.

We want to implement component related function at soc-component.c,
but, some of them need to care whole snd_soc_pcm_runtime (= rtd)
connected component.

Let's call component related function which need to care with
for_each_rtdcom() loop as snd_soc_pcm_component_xxx().
This patch adds new snd_soc_pcm_component_pcm_free() and use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k1c54czu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  1 +
 sound/soc/soc-component.c     | 14 ++++++++++++++
 sound/soc/soc-pcm.c           |  9 +--------
 3 files changed, 16 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index d3048ad06582..5d80b2eef525 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -382,5 +382,6 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
 int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
 			       struct vm_area_struct *vma);
 int snd_soc_pcm_component_new(struct snd_pcm *pcm);
+void snd_soc_pcm_component_free(struct snd_pcm *pcm);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index ff13d901bbab..79ffc2820ba9 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -545,3 +545,17 @@ int snd_soc_pcm_component_new(struct snd_pcm *pcm)
 
 	return 0;
 }
+
+void snd_soc_pcm_component_free(struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (component->driver->pcm_free)
+			component->driver->pcm_free(pcm);
+	}
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 955c49fd3bda..77c986fe08d0 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2805,17 +2805,10 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
 static void soc_pcm_private_free(struct snd_pcm *pcm)
 {
 	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
 
 	/* need to sync the delayed work before releasing resources */
 	flush_delayed_work(&rtd->delayed_work);
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (component->driver->pcm_free)
-			component->driver->pcm_free(pcm);
-	}
+	snd_soc_pcm_component_free(pcm);
 }
 
 /* create a new pcm */
-- 
cgit v1.2.3-59-g8ed1b


From 3ec3d7a3ff10692d4f8a2baa0ff18fe10a9b6ad4 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:20 +0900
Subject: ALSA: firewire-lib: add AMDTP domain structure to handle several isoc
 contexts

This commit adds 'struct amdtp_domain' structure. This structure
has list of instance of AMDTP stream to handle a couple of
isochronous contexts.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 22 ++++++++++++++++++++++
 sound/firewire/amdtp-stream.h |  7 +++++++
 2 files changed, 29 insertions(+)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 369e75e33120..02077696fa77 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -1136,3 +1136,25 @@ void amdtp_stream_pcm_abort(struct amdtp_stream *s)
 		snd_pcm_stop_xrun(pcm);
 }
 EXPORT_SYMBOL(amdtp_stream_pcm_abort);
+
+/**
+ * amdtp_domain_init - initialize an AMDTP domain structure
+ * @d: the AMDTP domain to initialize.
+ */
+int amdtp_domain_init(struct amdtp_domain *d)
+{
+	INIT_LIST_HEAD(&d->streams);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(amdtp_domain_init);
+
+/**
+ * amdtp_domain_destroy - destroy an AMDTP domain structure
+ * @d: the AMDTP domain to destroy.
+ */
+void amdtp_domain_destroy(struct amdtp_domain *d)
+{
+	WARN_ON(!list_empty(&d->streams));
+}
+EXPORT_SYMBOL_GPL(amdtp_domain_destroy);
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 50041fa884d9..2378db4b4195 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -267,4 +267,11 @@ static inline bool amdtp_stream_wait_callback(struct amdtp_stream *s,
 				  msecs_to_jiffies(timeout)) > 0;
 }
 
+struct amdtp_domain {
+	struct list_head streams;
+};
+
+int amdtp_domain_init(struct amdtp_domain *d);
+void amdtp_domain_destroy(struct amdtp_domain *d);
+
 #endif
-- 
cgit v1.2.3-59-g8ed1b


From 6261f90bdb0b605ffb158717ec7a03b1753aded5 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:21 +0900
Subject: ALSA: firewire-lib: add a kernel API to stop a couple of AMDTP
 streams in AMDTP domain

This commit adds a kernel API to stop a couple of isochronous contexts
for AMDTP streams. The API is not protected with any lock primitive.
Callers should use this with enough lock against concurrent access.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 16 ++++++++++++++++
 sound/firewire/amdtp-stream.h |  5 +++++
 2 files changed, 21 insertions(+)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 02077696fa77..31fc90f76443 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -1158,3 +1158,19 @@ void amdtp_domain_destroy(struct amdtp_domain *d)
 	WARN_ON(!list_empty(&d->streams));
 }
 EXPORT_SYMBOL_GPL(amdtp_domain_destroy);
+
+/**
+ * amdtp_domain_stop - stop sending packets for isoc context in the same domain.
+ * @d: the AMDTP domain to which the isoc contexts belong.
+ */
+void amdtp_domain_stop(struct amdtp_domain *d)
+{
+	struct amdtp_stream *s, *next;
+
+	list_for_each_entry_safe(s, next, &d->streams, list) {
+		list_del(&s->list);
+
+		amdtp_stream_stop(s);
+	}
+}
+EXPORT_SYMBOL_GPL(amdtp_domain_stop);
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 2378db4b4195..ab2a69180240 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -170,6 +170,9 @@ struct amdtp_stream {
 	/* For backends to process data blocks. */
 	void *protocol;
 	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
+
+	// For domain.
+	struct list_head list;
 };
 
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
@@ -274,4 +277,6 @@ struct amdtp_domain {
 int amdtp_domain_init(struct amdtp_domain *d);
 void amdtp_domain_destroy(struct amdtp_domain *d);
 
+void amdtp_domain_stop(struct amdtp_domain *d);
+
 #endif
-- 
cgit v1.2.3-59-g8ed1b


From 157a53eef6a016e6938d9fd266b85221fcd4eaa3 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:22 +0900
Subject: ALSA: firewire-lib: add a kernel API to add AMDTP stream into AMDTP
 domain

This commit adds a kernel API to insert AMDTP stream to list in AMDTP
domain.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 26 ++++++++++++++++++++++++++
 sound/firewire/amdtp-stream.h |  5 +++++
 2 files changed, 31 insertions(+)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 31fc90f76443..fa7989ee4769 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -1159,6 +1159,32 @@ void amdtp_domain_destroy(struct amdtp_domain *d)
 }
 EXPORT_SYMBOL_GPL(amdtp_domain_destroy);
 
+/**
+ * amdtp_domain_add_stream - register isoc context into the domain.
+ * @d: the AMDTP domain.
+ * @s: the AMDTP stream.
+ * @channel: the isochronous channel on the bus.
+ * @speed: firewire speed code.
+ */
+int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s,
+			    int channel, int speed)
+{
+	struct amdtp_stream *tmp;
+
+	list_for_each_entry(tmp, &d->streams, list) {
+		if (s == tmp)
+			return -EBUSY;
+	}
+
+	list_add(&s->list, &d->streams);
+
+	s->channel = channel;
+	s->speed = speed;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(amdtp_domain_add_stream);
+
 /**
  * amdtp_domain_stop - stop sending packets for isoc context in the same domain.
  * @d: the AMDTP domain to which the isoc contexts belong.
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index ab2a69180240..4b102fd7529d 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -172,6 +172,8 @@ struct amdtp_stream {
 	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 
 	// For domain.
+	int channel;
+	int speed;
 	struct list_head list;
 };
 
@@ -277,6 +279,9 @@ struct amdtp_domain {
 int amdtp_domain_init(struct amdtp_domain *d);
 void amdtp_domain_destroy(struct amdtp_domain *d);
 
+int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s,
+			    int channel, int speed);
+
 void amdtp_domain_stop(struct amdtp_domain *d);
 
 #endif
-- 
cgit v1.2.3-59-g8ed1b


From 9b4702b06c0e25abc612e6f02f3e25a51c684a01 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:23 +0900
Subject: ALSA: firewire-lib: add a kernel API to start AMDTP streams in AMDTP
 domain

This commit adds a kernel API to start a couple of isochronous contexts
for some AMDTP streams.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 24 ++++++++++++++++++++++++
 sound/firewire/amdtp-stream.h |  1 +
 2 files changed, 25 insertions(+)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index fa7989ee4769..158d210caea7 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -1185,6 +1185,30 @@ int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s,
 }
 EXPORT_SYMBOL_GPL(amdtp_domain_add_stream);
 
+/**
+ * amdtp_domain_start - start sending packets for isoc context in the domain.
+ * @d: the AMDTP domain.
+ */
+int amdtp_domain_start(struct amdtp_domain *d)
+{
+	struct amdtp_stream *s;
+	int err = 0;
+
+	list_for_each_entry(s, &d->streams, list) {
+		err = amdtp_stream_start(s, s->channel, s->speed);
+		if (err < 0)
+			break;
+	}
+
+	if (err < 0) {
+		list_for_each_entry(s, &d->streams, list)
+			amdtp_stream_stop(s);
+	}
+
+	return err;
+}
+EXPORT_SYMBOL_GPL(amdtp_domain_start);
+
 /**
  * amdtp_domain_stop - stop sending packets for isoc context in the same domain.
  * @d: the AMDTP domain to which the isoc contexts belong.
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 4b102fd7529d..15d471660a43 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -282,6 +282,7 @@ void amdtp_domain_destroy(struct amdtp_domain *d);
 int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s,
 			    int channel, int speed);
 
+int amdtp_domain_start(struct amdtp_domain *d);
 void amdtp_domain_stop(struct amdtp_domain *d);
 
 #endif
-- 
cgit v1.2.3-59-g8ed1b


From 94491c175d6bf4b838043a3fb5adabd731d8fbab Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:24 +0900
Subject: ALSA: fireworks: code refactoring for initialization/destruction of
 AMDTP streams

This commit is a preparation to support AMDTP domain.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/fireworks/fireworks_stream.c | 77 ++++++++++++++---------------
 1 file changed, 36 insertions(+), 41 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 385fc9686365..0f62c50055e9 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -8,8 +8,7 @@
 
 #define CALLBACK_TIMEOUT	100
 
-static int
-init_stream(struct snd_efw *efw, struct amdtp_stream *stream)
+static int init_stream(struct snd_efw *efw, struct amdtp_stream *stream)
 {
 	struct cmp_connection *conn;
 	enum cmp_direction c_dir;
@@ -28,14 +27,37 @@ init_stream(struct snd_efw *efw, struct amdtp_stream *stream)
 
 	err = cmp_connection_init(conn, efw->unit, c_dir, 0);
 	if (err < 0)
-		goto end;
+		return err;
 
 	err = amdtp_am824_init(stream, efw->unit, s_dir, CIP_BLOCKING);
 	if (err < 0) {
 		amdtp_stream_destroy(stream);
 		cmp_connection_destroy(conn);
+		return err;
 	}
-end:
+
+	if (stream == &efw->tx_stream) {
+		// Fireworks transmits NODATA packets with TAG0.
+		efw->tx_stream.flags |= CIP_EMPTY_WITH_TAG0;
+		// Fireworks has its own meaning for dbc.
+		efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT;
+		// Fireworks reset dbc at bus reset.
+		efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK;
+		// But Recent firmwares starts packets with non-zero dbc.
+		// Driver version 5.7.6 installs firmware version 5.7.3.
+		if (efw->is_fireworks3 &&
+		    (efw->firmware_version == 0x5070000 ||
+		     efw->firmware_version == 0x5070300 ||
+		     efw->firmware_version == 0x5080000))
+			efw->tx_stream.flags |= CIP_UNALIGHED_DBC;
+		// AudioFire9 always reports wrong dbs.
+		if (efw->is_af9)
+			efw->tx_stream.flags |= CIP_WRONG_DBS;
+		// Firmware version 5.5 reports fixed interval for dbc.
+		if (efw->firmware_version == 0x5050000)
+			efw->tx_stream.ctx_data.tx.dbc_interval = 8;
+	}
+
 	return err;
 }
 
@@ -83,22 +105,16 @@ static int start_stream(struct snd_efw *efw, struct amdtp_stream *stream,
 	return 0;
 }
 
-/*
- * This function should be called before starting the stream or after stopping
- * the streams.
- */
-static void
-destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream)
+// This function should be called before starting the stream or after stopping
+// the streams.
+static void destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream)
 {
-	struct cmp_connection *conn;
+	amdtp_stream_destroy(stream);
 
 	if (stream == &efw->tx_stream)
-		conn = &efw->out_conn;
+		cmp_connection_destroy(&efw->out_conn);
 	else
-		conn = &efw->in_conn;
-
-	amdtp_stream_destroy(stream);
-	cmp_connection_destroy(conn);
+		cmp_connection_destroy(&efw->in_conn);
 }
 
 static int
@@ -131,42 +147,21 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw)
 
 	err = init_stream(efw, &efw->tx_stream);
 	if (err < 0)
-		goto end;
-	/* Fireworks transmits NODATA packets with TAG0. */
-	efw->tx_stream.flags |= CIP_EMPTY_WITH_TAG0;
-	/* Fireworks has its own meaning for dbc. */
-	efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT;
-	/* Fireworks reset dbc at bus reset. */
-	efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK;
-	/*
-	 * But Recent firmwares starts packets with non-zero dbc.
-	 * Driver version 5.7.6 installs firmware version 5.7.3.
-	 */
-	if (efw->is_fireworks3 &&
-	    (efw->firmware_version == 0x5070000 ||
-	     efw->firmware_version == 0x5070300 ||
-	     efw->firmware_version == 0x5080000))
-		efw->tx_stream.flags |= CIP_UNALIGHED_DBC;
-	/* AudioFire9 always reports wrong dbs. */
-	if (efw->is_af9)
-		efw->tx_stream.flags |= CIP_WRONG_DBS;
-	/* Firmware version 5.5 reports fixed interval for dbc. */
-	if (efw->firmware_version == 0x5050000)
-		efw->tx_stream.ctx_data.tx.dbc_interval = 8;
+		return err;
 
 	err = init_stream(efw, &efw->rx_stream);
 	if (err < 0) {
 		destroy_stream(efw, &efw->tx_stream);
-		goto end;
+		return err;
 	}
 
-	/* set IEC61883 compliant mode (actually not fully compliant...) */
+	// set IEC61883 compliant mode (actually not fully compliant...).
 	err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883);
 	if (err < 0) {
 		destroy_stream(efw, &efw->tx_stream);
 		destroy_stream(efw, &efw->rx_stream);
 	}
-end:
+
 	return err;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 7eb7b18e9fc7c8dc3d3c186ec6b0cfa9147299f8 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:25 +0900
Subject: ALSA: fireworks: code refactoring for bus reset handler

This commit is a preparation to support AMDTP domain.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/fireworks/fireworks_stream.c | 13 +++++--------
 1 file changed, 5 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 0f62c50055e9..af340491dc43 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -299,14 +299,11 @@ void snd_efw_stream_stop_duplex(struct snd_efw *efw)
 
 void snd_efw_stream_update_duplex(struct snd_efw *efw)
 {
-	if (cmp_connection_update(&efw->out_conn) < 0 ||
-	    cmp_connection_update(&efw->in_conn) < 0) {
-		stop_stream(efw, &efw->rx_stream);
-		stop_stream(efw, &efw->tx_stream);
-	} else {
-		amdtp_stream_update(&efw->rx_stream);
-		amdtp_stream_update(&efw->tx_stream);
-	}
+	stop_stream(efw, &efw->rx_stream);
+	stop_stream(efw, &efw->tx_stream);
+
+	amdtp_stream_pcm_abort(&efw->rx_stream);
+	amdtp_stream_pcm_abort(&efw->tx_stream);
 }
 
 void snd_efw_stream_destroy_duplex(struct snd_efw *efw)
-- 
cgit v1.2.3-59-g8ed1b


From d79360ebe9c8668caf0e3932216f236fffbfa1b6 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:26 +0900
Subject: ALSA: firewire-digi00x: code refactoring for
 initialization/destruction of AMDTP stream

This commit is a preparation to support AMDTP domain.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/digi00x/digi00x-stream.c | 65 +++++++++++++++++++++------------
 1 file changed, 42 insertions(+), 23 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c
index 3e77dbd3ee22..cff193f00a97 100644
--- a/sound/firewire/digi00x/digi00x-stream.c
+++ b/sound/firewire/digi00x/digi00x-stream.c
@@ -218,43 +218,62 @@ static int keep_resources(struct snd_dg00x *dg00x, struct amdtp_stream *stream,
 				fw_parent_device(dg00x->unit)->max_speed);
 }
 
-int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
+static int init_stream(struct snd_dg00x *dg00x, struct amdtp_stream *s)
 {
+	struct fw_iso_resources *resources;
+	enum amdtp_stream_direction dir;
 	int err;
 
-	/* For out-stream. */
-	err = fw_iso_resources_init(&dg00x->rx_resources, dg00x->unit);
+	if (s == &dg00x->tx_stream) {
+		resources = &dg00x->tx_resources;
+		dir = AMDTP_IN_STREAM;
+	} else {
+		resources = &dg00x->rx_resources;
+		dir = AMDTP_OUT_STREAM;
+	}
+
+	err = fw_iso_resources_init(resources, dg00x->unit);
 	if (err < 0)
-		goto error;
-	err = amdtp_dot_init(&dg00x->rx_stream, dg00x->unit, AMDTP_OUT_STREAM);
+		return err;
+
+	err = amdtp_dot_init(s, dg00x->unit, dir);
 	if (err < 0)
-		goto error;
+		fw_iso_resources_destroy(resources);
+
+	return err;
+}
 
-	/* For in-stream. */
-	err = fw_iso_resources_init(&dg00x->tx_resources, dg00x->unit);
+static void destroy_stream(struct snd_dg00x *dg00x, struct amdtp_stream *s)
+{
+	amdtp_stream_destroy(s);
+
+	if (s == &dg00x->tx_stream)
+		fw_iso_resources_destroy(&dg00x->tx_resources);
+	else
+		fw_iso_resources_destroy(&dg00x->rx_resources);
+}
+
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
+{
+	int err;
+
+	err = init_stream(dg00x, &dg00x->rx_stream);
 	if (err < 0)
-		goto error;
-	err = amdtp_dot_init(&dg00x->tx_stream, dg00x->unit, AMDTP_IN_STREAM);
+		return err;
+
+	err = init_stream(dg00x, &dg00x->tx_stream);
 	if (err < 0)
-		goto error;
+		destroy_stream(dg00x, &dg00x->rx_stream);
 
-	return 0;
-error:
-	snd_dg00x_stream_destroy_duplex(dg00x);
 	return err;
 }
 
-/*
- * This function should be called before starting streams or after stopping
- * streams.
- */
+// This function should be called before starting streams or after stopping
+// streams.
 void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x)
 {
-	amdtp_stream_destroy(&dg00x->rx_stream);
-	fw_iso_resources_destroy(&dg00x->rx_resources);
-
-	amdtp_stream_destroy(&dg00x->tx_stream);
-	fw_iso_resources_destroy(&dg00x->tx_resources);
+	destroy_stream(dg00x, &dg00x->rx_stream);
+	destroy_stream(dg00x, &dg00x->tx_stream);
 }
 
 int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate)
-- 
cgit v1.2.3-59-g8ed1b


From 5f9625a5ba5dcf1815f123fced65f289b558aca4 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:27 +0900
Subject: ALSA: firewire-tascam: code refactoring for
 initialization/destruction of AMDTP stream

This commit is a preparation to support AMDTP domain.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/tascam/tascam-stream.c | 71 ++++++++++++++++++++++-------------
 1 file changed, 44 insertions(+), 27 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
index e852e46ebe6f..d5e77036e0ee 100644
--- a/sound/firewire/tascam/tascam-stream.c
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -287,38 +287,60 @@ static int keep_resources(struct snd_tscm *tscm, unsigned int rate,
 				fw_parent_device(tscm->unit)->max_speed);
 }
 
-int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
+static int init_stream(struct snd_tscm *tscm, struct amdtp_stream *s)
 {
+	struct fw_iso_resources *resources;
+	enum amdtp_stream_direction dir;
 	unsigned int pcm_channels;
 	int err;
 
-	/* For out-stream. */
-	err = fw_iso_resources_init(&tscm->rx_resources, tscm->unit);
-	if (err < 0)
-		return err;
-	pcm_channels = tscm->spec->pcm_playback_analog_channels;
+	if (s == &tscm->tx_stream) {
+		resources = &tscm->tx_resources;
+		dir = AMDTP_IN_STREAM;
+		pcm_channels = tscm->spec->pcm_capture_analog_channels;
+	} else {
+		resources = &tscm->rx_resources;
+		dir = AMDTP_OUT_STREAM;
+		pcm_channels = tscm->spec->pcm_playback_analog_channels;
+	}
+
 	if (tscm->spec->has_adat)
 		pcm_channels += 8;
 	if (tscm->spec->has_spdif)
 		pcm_channels += 2;
-	err = amdtp_tscm_init(&tscm->rx_stream, tscm->unit, AMDTP_OUT_STREAM,
-			      pcm_channels);
+
+	err = fw_iso_resources_init(resources, tscm->unit);
 	if (err < 0)
 		return err;
 
-	/* For in-stream. */
-	err = fw_iso_resources_init(&tscm->tx_resources, tscm->unit);
+	err = amdtp_tscm_init(s, tscm->unit, dir, pcm_channels);
+	if (err < 0)
+		fw_iso_resources_free(resources);
+
+	return err;
+}
+
+static void destroy_stream(struct snd_tscm *tscm, struct amdtp_stream *s)
+{
+	amdtp_stream_destroy(s);
+
+	if (s == &tscm->tx_stream)
+		fw_iso_resources_destroy(&tscm->tx_resources);
+	else
+		fw_iso_resources_destroy(&tscm->rx_resources);
+}
+
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
+{
+	int err;
+
+	err = init_stream(tscm, &tscm->tx_stream);
 	if (err < 0)
 		return err;
-	pcm_channels = tscm->spec->pcm_capture_analog_channels;
-	if (tscm->spec->has_adat)
-		pcm_channels += 8;
-	if (tscm->spec->has_spdif)
-		pcm_channels += 2;
-	err = amdtp_tscm_init(&tscm->tx_stream, tscm->unit, AMDTP_IN_STREAM,
-			      pcm_channels);
+
+	err = init_stream(tscm, &tscm->rx_stream);
 	if (err < 0)
-		amdtp_stream_destroy(&tscm->rx_stream);
+		destroy_stream(tscm, &tscm->tx_stream);
 
 	return err;
 }
@@ -333,17 +355,12 @@ void snd_tscm_stream_update_duplex(struct snd_tscm *tscm)
 	amdtp_stream_stop(&tscm->rx_stream);
 }
 
-/*
- * This function should be called before starting streams or after stopping
- * streams.
- */
+// This function should be called before starting streams or after stopping
+// streams.
 void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm)
 {
-	amdtp_stream_destroy(&tscm->rx_stream);
-	amdtp_stream_destroy(&tscm->tx_stream);
-
-	fw_iso_resources_destroy(&tscm->rx_resources);
-	fw_iso_resources_destroy(&tscm->tx_resources);
+	destroy_stream(tscm, &tscm->rx_stream);
+	destroy_stream(tscm, &tscm->tx_stream);
 }
 
 int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate)
-- 
cgit v1.2.3-59-g8ed1b


From 39e522a5a45fe805a2925aaddc30a4b2c6ded642 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:28 +0900
Subject: ALSA: firewire-motu: code refactoring for initialization/destruction
 of AMDTP stream

This commit is a preparation to support AMDTP domain.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/motu/motu-stream.c | 48 ++++++++++++++++-----------------------
 1 file changed, 19 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c
index 2bbb335e8de1..cc9f34426a47 100644
--- a/sound/firewire/motu/motu-stream.c
+++ b/sound/firewire/motu/motu-stream.c
@@ -300,62 +300,52 @@ void snd_motu_stream_stop_duplex(struct snd_motu *motu)
 	}
 }
 
-static int init_stream(struct snd_motu *motu, enum amdtp_stream_direction dir)
+static int init_stream(struct snd_motu *motu, struct amdtp_stream *s)
 {
-	int err;
-	struct amdtp_stream *stream;
 	struct fw_iso_resources *resources;
+	enum amdtp_stream_direction dir;
+	int err;
 
-	if (dir == AMDTP_IN_STREAM) {
-		stream = &motu->tx_stream;
+	if (s == &motu->tx_stream) {
 		resources = &motu->tx_resources;
+		dir = AMDTP_IN_STREAM;
 	} else {
-		stream = &motu->rx_stream;
 		resources = &motu->rx_resources;
+		dir = AMDTP_OUT_STREAM;
 	}
 
 	err = fw_iso_resources_init(resources, motu->unit);
 	if (err < 0)
 		return err;
 
-	err = amdtp_motu_init(stream, motu->unit, dir, motu->spec->protocol);
-	if (err < 0) {
-		amdtp_stream_destroy(stream);
+	err = amdtp_motu_init(s, motu->unit, dir, motu->spec->protocol);
+	if (err < 0)
 		fw_iso_resources_destroy(resources);
-	}
 
 	return err;
 }
 
-static void destroy_stream(struct snd_motu *motu,
-			   enum amdtp_stream_direction dir)
+static void destroy_stream(struct snd_motu *motu, struct amdtp_stream *s)
 {
-	struct amdtp_stream *stream;
-	struct fw_iso_resources *resources;
+	amdtp_stream_destroy(s);
 
-	if (dir == AMDTP_IN_STREAM) {
-		stream = &motu->tx_stream;
-		resources = &motu->tx_resources;
-	} else {
-		stream = &motu->rx_stream;
-		resources = &motu->rx_resources;
-	}
-
-	amdtp_stream_destroy(stream);
-	fw_iso_resources_destroy(resources);
+	if (s == &motu->tx_stream)
+		fw_iso_resources_destroy(&motu->tx_resources);
+	else
+		fw_iso_resources_destroy(&motu->rx_resources);
 }
 
 int snd_motu_stream_init_duplex(struct snd_motu *motu)
 {
 	int err;
 
-	err = init_stream(motu, AMDTP_IN_STREAM);
+	err = init_stream(motu, &motu->tx_stream);
 	if (err < 0)
 		return err;
 
-	err = init_stream(motu, AMDTP_OUT_STREAM);
+	err = init_stream(motu, &motu->rx_stream);
 	if (err < 0)
-		destroy_stream(motu, AMDTP_IN_STREAM);
+		destroy_stream(motu, &motu->tx_stream);
 
 	return err;
 }
@@ -366,8 +356,8 @@ int snd_motu_stream_init_duplex(struct snd_motu *motu)
  */
 void snd_motu_stream_destroy_duplex(struct snd_motu *motu)
 {
-	destroy_stream(motu, AMDTP_IN_STREAM);
-	destroy_stream(motu, AMDTP_OUT_STREAM);
+	destroy_stream(motu, &motu->rx_stream);
+	destroy_stream(motu, &motu->tx_stream);
 
 	motu->substreams_counter = 0;
 }
-- 
cgit v1.2.3-59-g8ed1b


From 42355abb9c33503f7b5889d71e1432fabdf2fd7f Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:29 +0900
Subject: ALSA: fireface: code refactoring for initialization/destruction of
 AMDTP stream

This commit is a preparation to support AMDTP domain.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/fireface/ff-stream.c | 39 ++++++++++++++++++-------------------
 1 file changed, 19 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c
index 4208b8004d1a..e4710204f481 100644
--- a/sound/firewire/fireface/ff-stream.c
+++ b/sound/firewire/fireface/ff-stream.c
@@ -39,54 +39,53 @@ static inline void finish_session(struct snd_ff *ff)
 	ff->spec->protocol->switch_fetching_mode(ff, false);
 }
 
-static int init_stream(struct snd_ff *ff, enum amdtp_stream_direction dir)
+static int init_stream(struct snd_ff *ff, struct amdtp_stream *s)
 {
-	int err;
 	struct fw_iso_resources *resources;
-	struct amdtp_stream *stream;
+	enum amdtp_stream_direction dir;
+	int err;
 
-	if (dir == AMDTP_IN_STREAM) {
+	if (s == &ff->tx_stream) {
 		resources = &ff->tx_resources;
-		stream = &ff->tx_stream;
+		dir = AMDTP_IN_STREAM;
 	} else {
 		resources = &ff->rx_resources;
-		stream = &ff->rx_stream;
+		dir = AMDTP_OUT_STREAM;
 	}
 
 	err = fw_iso_resources_init(resources, ff->unit);
 	if (err < 0)
 		return err;
 
-	err = amdtp_ff_init(stream, ff->unit, dir);
+	err = amdtp_ff_init(s, ff->unit, dir);
 	if (err < 0)
 		fw_iso_resources_destroy(resources);
 
 	return err;
 }
 
-static void destroy_stream(struct snd_ff *ff, enum amdtp_stream_direction dir)
+static void destroy_stream(struct snd_ff *ff, struct amdtp_stream *s)
 {
-	if (dir == AMDTP_IN_STREAM) {
-		amdtp_stream_destroy(&ff->tx_stream);
+	amdtp_stream_destroy(s);
+
+	if (s == &ff->tx_stream)
 		fw_iso_resources_destroy(&ff->tx_resources);
-	} else {
-		amdtp_stream_destroy(&ff->rx_stream);
+	else
 		fw_iso_resources_destroy(&ff->rx_resources);
-	}
 }
 
 int snd_ff_stream_init_duplex(struct snd_ff *ff)
 {
 	int err;
 
-	err = init_stream(ff, AMDTP_OUT_STREAM);
+	err = init_stream(ff, &ff->rx_stream);
 	if (err < 0)
-		goto end;
+		return err;
 
-	err = init_stream(ff, AMDTP_IN_STREAM);
+	err = init_stream(ff, &ff->tx_stream);
 	if (err < 0)
-		destroy_stream(ff, AMDTP_OUT_STREAM);
-end:
+		destroy_stream(ff, &ff->rx_stream);
+
 	return err;
 }
 
@@ -96,8 +95,8 @@ end:
  */
 void snd_ff_stream_destroy_duplex(struct snd_ff *ff)
 {
-	destroy_stream(ff, AMDTP_IN_STREAM);
-	destroy_stream(ff, AMDTP_OUT_STREAM);
+	destroy_stream(ff, &ff->rx_stream);
+	destroy_stream(ff, &ff->tx_stream);
 }
 
 int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate)
-- 
cgit v1.2.3-59-g8ed1b


From b0db4d512941455894e7a13f71594b7d4638b94b Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:30 +0900
Subject: ALSA: bebob: support AMDTP domain

This commit adds AMDTP domain support for ALSA bebob driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/bebob/bebob.h        |  2 ++
 sound/firewire/bebob/bebob_stream.c | 62 ++++++++++++++++---------------------
 2 files changed, 29 insertions(+), 35 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index 9e0b689fe34a..356d6ba60959 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -115,6 +115,8 @@ struct snd_bebob {
 
 	/* For BeBoB version quirk. */
 	unsigned int version;
+
+	struct amdtp_domain domain;
 };
 
 static inline int
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 334dc7c96e1d..73fee991bd75 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -445,10 +445,9 @@ start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream)
 			goto end;
 	}
 
-	/* start amdtp stream */
-	err = amdtp_stream_start(stream,
-				 conn->resources.channel,
-				 conn->speed);
+	// start amdtp stream.
+	err = amdtp_domain_add_stream(&bebob->domain, stream,
+				      conn->resources.channel, conn->speed);
 end:
 	return err;
 }
@@ -523,7 +522,13 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
 		return err;
 	}
 
-	return 0;
+	err = amdtp_domain_init(&bebob->domain);
+	if (err < 0) {
+		destroy_stream(bebob, &bebob->tx_stream);
+		destroy_stream(bebob, &bebob->rx_stream);
+	}
+
+	return err;
 }
 
 static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream,
@@ -566,9 +571,7 @@ int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate)
 	if (rate == 0)
 		rate = curr_rate;
 	if (curr_rate != rate) {
-		amdtp_stream_stop(&bebob->tx_stream);
-		amdtp_stream_stop(&bebob->rx_stream);
-
+		amdtp_domain_stop(&bebob->domain);
 		break_both_connections(bebob);
 
 		cmp_connection_release(&bebob->out_conn);
@@ -620,9 +623,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob)
 	// packet queueing error or detecting discontinuity
 	if (amdtp_streaming_error(&bebob->rx_stream) ||
 	    amdtp_streaming_error(&bebob->tx_stream)) {
-		amdtp_stream_stop(&bebob->rx_stream);
-		amdtp_stream_stop(&bebob->tx_stream);
-
+		amdtp_domain_stop(&bebob->domain);
 		break_both_connections(bebob);
 	}
 
@@ -640,11 +641,16 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob)
 			return err;
 
 		err = start_stream(bebob, &bebob->rx_stream);
-		if (err < 0) {
-			dev_err(&bebob->unit->device,
-				"fail to run AMDTP master stream:%d\n", err);
+		if (err < 0)
+			goto error;
+
+		err = start_stream(bebob, &bebob->tx_stream);
+		if (err < 0)
+			goto error;
+
+		err = amdtp_domain_start(&bebob->domain);
+		if (err < 0)
 			goto error;
-		}
 
 		// NOTE:
 		// The firmware customized by M-Audio uses these commands to
@@ -660,21 +666,8 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob)
 		}
 
 		if (!amdtp_stream_wait_callback(&bebob->rx_stream,
-						CALLBACK_TIMEOUT)) {
-			err = -ETIMEDOUT;
-			goto error;
-		}
-	}
-
-	if (!amdtp_stream_running(&bebob->tx_stream)) {
-		err = start_stream(bebob, &bebob->tx_stream);
-		if (err < 0) {
-			dev_err(&bebob->unit->device,
-				"fail to run AMDTP slave stream:%d\n", err);
-			goto error;
-		}
-
-		if (!amdtp_stream_wait_callback(&bebob->tx_stream,
+						CALLBACK_TIMEOUT) ||
+		    !amdtp_stream_wait_callback(&bebob->tx_stream,
 						CALLBACK_TIMEOUT)) {
 			err = -ETIMEDOUT;
 			goto error;
@@ -683,8 +676,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob)
 
 	return 0;
 error:
-	amdtp_stream_stop(&bebob->tx_stream);
-	amdtp_stream_stop(&bebob->rx_stream);
+	amdtp_domain_stop(&bebob->domain);
 	break_both_connections(bebob);
 	return err;
 }
@@ -692,9 +684,7 @@ error:
 void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob)
 {
 	if (bebob->substreams_counter == 0) {
-		amdtp_stream_stop(&bebob->rx_stream);
-		amdtp_stream_stop(&bebob->tx_stream);
-
+		amdtp_domain_stop(&bebob->domain);
 		break_both_connections(bebob);
 
 		cmp_connection_release(&bebob->out_conn);
@@ -708,6 +698,8 @@ void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob)
  */
 void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob)
 {
+	amdtp_domain_destroy(&bebob->domain);
+
 	destroy_stream(bebob, &bebob->tx_stream);
 	destroy_stream(bebob, &bebob->rx_stream);
 }
-- 
cgit v1.2.3-59-g8ed1b


From db40eeb2469fef4efe26789ea2596372ee43ae2f Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:31 +0900
Subject: ALSA: fireworks: support AMDTP domain

This commit adds AMDTP domain support for ALSA fireworks driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/fireworks/fireworks.h        |  2 +
 sound/firewire/fireworks/fireworks_stream.c | 92 +++++++++++++++--------------
 2 files changed, 51 insertions(+), 43 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 31efd4b53b4f..4cda297f8438 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -107,6 +107,8 @@ struct snd_efw {
 	u8 *resp_buf;
 	u8 *pull_ptr;
 	u8 *push_ptr;
+
+	struct amdtp_domain domain;
 };
 
 int snd_efw_transaction_cmd(struct fw_unit *unit,
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index af340491dc43..f2de304d2f26 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -61,17 +61,6 @@ static int init_stream(struct snd_efw *efw, struct amdtp_stream *stream)
 	return err;
 }
 
-static void
-stop_stream(struct snd_efw *efw, struct amdtp_stream *stream)
-{
-	amdtp_stream_stop(stream);
-
-	if (stream == &efw->tx_stream)
-		cmp_connection_break(&efw->out_conn);
-	else
-		cmp_connection_break(&efw->in_conn);
-}
-
 static int start_stream(struct snd_efw *efw, struct amdtp_stream *stream,
 			unsigned int rate)
 {
@@ -89,19 +78,13 @@ static int start_stream(struct snd_efw *efw, struct amdtp_stream *stream,
 		return err;
 
 	// Start amdtp stream.
-	err = amdtp_stream_start(stream, conn->resources.channel, conn->speed);
+	err = amdtp_domain_add_stream(&efw->domain, stream,
+				      conn->resources.channel, conn->speed);
 	if (err < 0) {
 		cmp_connection_break(conn);
 		return err;
 	}
 
-	// Wait first callback.
-	if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) {
-		amdtp_stream_stop(stream);
-		cmp_connection_break(conn);
-		return -ETIMEDOUT;
-	}
-
 	return 0;
 }
 
@@ -155,6 +138,13 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw)
 		return err;
 	}
 
+	err = amdtp_domain_init(&efw->domain);
+	if (err < 0) {
+		destroy_stream(efw, &efw->tx_stream);
+		destroy_stream(efw, &efw->rx_stream);
+		return err;
+	}
+
 	// set IEC61883 compliant mode (actually not fully compliant...).
 	err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883);
 	if (err < 0) {
@@ -209,8 +199,10 @@ int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate)
 	if (rate == 0)
 		rate = curr_rate;
 	if (rate != curr_rate) {
-		stop_stream(efw, &efw->tx_stream);
-		stop_stream(efw, &efw->rx_stream);
+		amdtp_domain_stop(&efw->domain);
+
+		cmp_connection_break(&efw->out_conn);
+		cmp_connection_break(&efw->in_conn);
 
 		cmp_connection_release(&efw->out_conn);
 		cmp_connection_release(&efw->in_conn);
@@ -250,47 +242,57 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw)
 	if (efw->substreams_counter == 0)
 		return -EIO;
 
-	err = snd_efw_command_get_sampling_rate(efw, &rate);
-	if (err < 0)
-		return err;
-
 	if (amdtp_streaming_error(&efw->rx_stream) ||
 	    amdtp_streaming_error(&efw->tx_stream)) {
-		stop_stream(efw, &efw->rx_stream);
-		stop_stream(efw, &efw->tx_stream);
+		amdtp_domain_stop(&efw->domain);
+		cmp_connection_break(&efw->out_conn);
+		cmp_connection_break(&efw->in_conn);
 	}
 
-	/* master should be always running */
+	err = snd_efw_command_get_sampling_rate(efw, &rate);
+	if (err < 0)
+		return err;
+
 	if (!amdtp_stream_running(&efw->rx_stream)) {
 		err = start_stream(efw, &efw->rx_stream, rate);
-		if (err < 0) {
-			dev_err(&efw->unit->device,
-				"fail to start AMDTP master stream:%d\n", err);
+		if (err < 0)
 			goto error;
-		}
-	}
 
-	if (!amdtp_stream_running(&efw->tx_stream)) {
 		err = start_stream(efw, &efw->tx_stream, rate);
-		if (err < 0) {
-			dev_err(&efw->unit->device,
-				"fail to start AMDTP slave stream:%d\n", err);
+		if (err < 0)
+			goto error;
+
+		err = amdtp_domain_start(&efw->domain);
+		if (err < 0)
+			goto error;
+
+		// Wait first callback.
+		if (!amdtp_stream_wait_callback(&efw->rx_stream,
+						CALLBACK_TIMEOUT) ||
+		    !amdtp_stream_wait_callback(&efw->tx_stream,
+						CALLBACK_TIMEOUT)) {
+			err = -ETIMEDOUT;
 			goto error;
 		}
 	}
 
 	return 0;
 error:
-	stop_stream(efw, &efw->rx_stream);
-	stop_stream(efw, &efw->tx_stream);
+	amdtp_domain_stop(&efw->domain);
+
+	cmp_connection_break(&efw->out_conn);
+	cmp_connection_break(&efw->in_conn);
+
 	return err;
 }
 
 void snd_efw_stream_stop_duplex(struct snd_efw *efw)
 {
 	if (efw->substreams_counter == 0) {
-		stop_stream(efw, &efw->tx_stream);
-		stop_stream(efw, &efw->rx_stream);
+		amdtp_domain_stop(&efw->domain);
+
+		cmp_connection_break(&efw->out_conn);
+		cmp_connection_break(&efw->in_conn);
 
 		cmp_connection_release(&efw->out_conn);
 		cmp_connection_release(&efw->in_conn);
@@ -299,8 +301,10 @@ void snd_efw_stream_stop_duplex(struct snd_efw *efw)
 
 void snd_efw_stream_update_duplex(struct snd_efw *efw)
 {
-	stop_stream(efw, &efw->rx_stream);
-	stop_stream(efw, &efw->tx_stream);
+	amdtp_domain_stop(&efw->domain);
+
+	cmp_connection_break(&efw->out_conn);
+	cmp_connection_break(&efw->in_conn);
 
 	amdtp_stream_pcm_abort(&efw->rx_stream);
 	amdtp_stream_pcm_abort(&efw->tx_stream);
@@ -308,6 +312,8 @@ void snd_efw_stream_update_duplex(struct snd_efw *efw)
 
 void snd_efw_stream_destroy_duplex(struct snd_efw *efw)
 {
+	amdtp_domain_destroy(&efw->domain);
+
 	destroy_stream(efw, &efw->rx_stream);
 	destroy_stream(efw, &efw->tx_stream);
 }
-- 
cgit v1.2.3-59-g8ed1b


From ac5d77864cfcb0f034221c2b3dac828ae24c722e Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:32 +0900
Subject: ALSA: oxfw: support AMDTP domain

This commit adds AMDTP domain support for ALSA oxfw driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/oxfw/oxfw-stream.c | 72 +++++++++++++++++++++++----------------
 sound/firewire/oxfw/oxfw.h        |  2 ++
 2 files changed, 45 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 74c972d25c66..0318dc4dcc55 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -114,19 +114,13 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream)
 	if (err < 0)
 		return err;
 
-	err = amdtp_stream_start(stream, conn->resources.channel, conn->speed);
+	err = amdtp_domain_add_stream(&oxfw->domain, stream,
+				      conn->resources.channel, conn->speed);
 	if (err < 0) {
 		cmp_connection_break(conn);
 		return err;
 	}
 
-	// Wait first packet.
-	if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) {
-		amdtp_stream_stop(stream);
-		cmp_connection_break(conn);
-		return -ETIMEDOUT;
-	}
-
 	return 0;
 }
 
@@ -280,12 +274,12 @@ int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw,
 		pcm_channels = formation.pcm;
 	}
 	if (formation.rate != rate || formation.pcm != pcm_channels) {
-		amdtp_stream_stop(&oxfw->rx_stream);
+		amdtp_domain_stop(&oxfw->domain);
+
 		cmp_connection_break(&oxfw->in_conn);
 		cmp_connection_release(&oxfw->in_conn);
 
 		if (oxfw->has_output) {
-			amdtp_stream_stop(&oxfw->tx_stream);
 			cmp_connection_break(&oxfw->out_conn);
 			cmp_connection_release(&oxfw->out_conn);
 		}
@@ -325,55 +319,66 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw)
 
 	if (amdtp_streaming_error(&oxfw->rx_stream) ||
 	    amdtp_streaming_error(&oxfw->tx_stream)) {
-		amdtp_stream_stop(&oxfw->rx_stream);
-		cmp_connection_break(&oxfw->in_conn);
+		amdtp_domain_stop(&oxfw->domain);
 
-		if (oxfw->has_output) {
-			amdtp_stream_stop(&oxfw->tx_stream);
+		cmp_connection_break(&oxfw->in_conn);
+		if (oxfw->has_output)
 			cmp_connection_break(&oxfw->out_conn);
-		}
 	}
 
 	if (!amdtp_stream_running(&oxfw->rx_stream)) {
 		err = start_stream(oxfw, &oxfw->rx_stream);
 		if (err < 0) {
 			dev_err(&oxfw->unit->device,
-				"fail to start rx stream: %d\n", err);
+				"fail to prepare rx stream: %d\n", err);
 			goto error;
 		}
-	}
 
-	if (oxfw->has_output) {
-		if (!amdtp_stream_running(&oxfw->tx_stream)) {
+		if (oxfw->has_output &&
+		    !amdtp_stream_running(&oxfw->tx_stream)) {
 			err = start_stream(oxfw, &oxfw->tx_stream);
 			if (err < 0) {
 				dev_err(&oxfw->unit->device,
-					"fail to start tx stream: %d\n", err);
+					"fail to prepare tx stream: %d\n", err);
 				goto error;
 			}
 		}
+
+		err = amdtp_domain_start(&oxfw->domain);
+		if (err < 0)
+			goto error;
+
+		// Wait first packet.
+		if (!amdtp_stream_wait_callback(&oxfw->rx_stream,
+						CALLBACK_TIMEOUT) ||
+		    !amdtp_stream_wait_callback(&oxfw->tx_stream,
+						CALLBACK_TIMEOUT)) {
+			err = -ETIMEDOUT;
+			goto error;
+		}
+
 	}
 
 	return 0;
 error:
-	amdtp_stream_stop(&oxfw->rx_stream);
+	amdtp_domain_stop(&oxfw->domain);
+
 	cmp_connection_break(&oxfw->in_conn);
-	if (oxfw->has_output) {
-		amdtp_stream_stop(&oxfw->tx_stream);
+	if (oxfw->has_output)
 		cmp_connection_break(&oxfw->out_conn);
-	}
+
 	return err;
 }
 
 void snd_oxfw_stream_stop_duplex(struct snd_oxfw *oxfw)
 {
 	if (oxfw->substreams_count == 0) {
-		amdtp_stream_stop(&oxfw->rx_stream);
+		amdtp_domain_stop(&oxfw->domain);
+
 		cmp_connection_break(&oxfw->in_conn);
 		cmp_connection_release(&oxfw->in_conn);
 
 		if (oxfw->has_output) {
-			amdtp_stream_stop(&oxfw->tx_stream);
 			cmp_connection_break(&oxfw->out_conn);
 			cmp_connection_release(&oxfw->out_conn);
 		}
@@ -409,13 +414,22 @@ int snd_oxfw_stream_init_duplex(struct snd_oxfw *oxfw)
 		}
 	}
 
-	return 0;
+	err = amdtp_domain_init(&oxfw->domain);
+	if (err < 0) {
+		destroy_stream(oxfw, &oxfw->rx_stream);
+		if (oxfw->has_output)
+			destroy_stream(oxfw, &oxfw->tx_stream);
+	}
+
+	return err;
 }
 
 // This function should be called before starting the stream or after stopping
 // the streams.
 void snd_oxfw_stream_destroy_duplex(struct snd_oxfw *oxfw)
 {
+	amdtp_domain_destroy(&oxfw->domain);
+
 	destroy_stream(oxfw, &oxfw->rx_stream);
 
 	if (oxfw->has_output)
@@ -424,13 +438,13 @@ void snd_oxfw_stream_destroy_duplex(struct snd_oxfw *oxfw)
 
 void snd_oxfw_stream_update_duplex(struct snd_oxfw *oxfw)
 {
-	amdtp_stream_stop(&oxfw->rx_stream);
+	amdtp_domain_stop(&oxfw->domain);
+
 	cmp_connection_break(&oxfw->in_conn);
 
 	amdtp_stream_pcm_abort(&oxfw->rx_stream);
 
 	if (oxfw->has_output) {
-		amdtp_stream_stop(&oxfw->tx_stream);
 		cmp_connection_break(&oxfw->out_conn);
 
 		amdtp_stream_pcm_abort(&oxfw->tx_stream);
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index cb69ab87bb14..c9627b8c5d6e 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -63,6 +63,8 @@ struct snd_oxfw {
 
 	const struct ieee1394_device_id *entry;
 	void *spec;
+
+	struct amdtp_domain domain;
 };
 
 /*
-- 
cgit v1.2.3-59-g8ed1b


From e9f21129b8d84debf53e06f73d340d5af136270f Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:33 +0900
Subject: ALSA: dice: support AMDTP domain

This commit adds AMDTP domain support for ALSA dice driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/dice/dice-stream.c | 32 +++++++++++++++++++++++++-------
 sound/firewire/dice/dice.h        |  2 ++
 2 files changed, 27 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index a9f0c77734c3..af8c5a2c28f3 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -154,14 +154,10 @@ static void stop_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
 	for (i = 0; i < params->count; i++) {
 		reg = cpu_to_be32((u32)-1);
 		if (dir == AMDTP_IN_STREAM) {
-			amdtp_stream_stop(&dice->tx_stream[i]);
-
 			snd_dice_transaction_write_tx(dice,
 					params->size * i + TX_ISOCHRONOUS,
 					&reg, sizeof(reg));
 		} else {
-			amdtp_stream_stop(&dice->rx_stream[i]);
-
 			snd_dice_transaction_write_rx(dice,
 					params->size * i + RX_ISOCHRONOUS,
 					&reg, sizeof(reg));
@@ -297,10 +293,11 @@ int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate)
 	if (dice->substreams_counter == 0 || curr_rate != rate) {
 		struct reg_params tx_params, rx_params;
 
+		amdtp_domain_stop(&dice->domain);
+
 		err = get_register_params(dice, &tx_params, &rx_params);
 		if (err < 0)
 			return err;
-
 		finish_session(dice, &tx_params, &rx_params);
 
 		release_resources(dice);
@@ -377,7 +374,8 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
 				return err;
 		}
 
-		err = amdtp_stream_start(stream, resources->channel, max_speed);
+		err = amdtp_domain_add_stream(&dice->domain, stream,
+					      resources->channel, max_speed);
 		if (err < 0)
 			return err;
 	}
@@ -410,6 +408,7 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice)
 	for (i = 0; i < MAX_STREAMS; ++i) {
 		if (amdtp_streaming_error(&dice->tx_stream[i]) ||
 		    amdtp_streaming_error(&dice->rx_stream[i])) {
+			amdtp_domain_stop(&dice->domain);
 			finish_session(dice, &tx_params, &rx_params);
 			break;
 		}
@@ -456,6 +455,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice)
 			goto error;
 		}
 
+		err = amdtp_domain_start(&dice->domain);
+		if (err < 0)
+			goto error;
+
 		for (i = 0; i < MAX_STREAMS; i++) {
 			if ((i < tx_params.count &&
 			    !amdtp_stream_wait_callback(&dice->tx_stream[i],
@@ -471,6 +474,7 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice)
 
 	return 0;
 error:
+	amdtp_domain_stop(&dice->domain);
 	finish_session(dice, &tx_params, &rx_params);
 	return err;
 }
@@ -485,8 +489,10 @@ void snd_dice_stream_stop_duplex(struct snd_dice *dice)
 	struct reg_params tx_params, rx_params;
 
 	if (dice->substreams_counter == 0) {
-		if (get_register_params(dice, &tx_params, &rx_params) >= 0)
+		if (get_register_params(dice, &tx_params, &rx_params) >= 0) {
+			amdtp_domain_stop(&dice->domain);
 			finish_session(dice, &tx_params, &rx_params);
+		}
 
 		release_resources(dice);
 	}
@@ -567,6 +573,14 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice)
 			break;
 		}
 	}
+
+	err = amdtp_domain_init(&dice->domain);
+	if (err < 0) {
+		for (i = 0; i < MAX_STREAMS; ++i) {
+			destroy_stream(dice, AMDTP_OUT_STREAM, i);
+			destroy_stream(dice, AMDTP_IN_STREAM, i);
+		}
+	}
 end:
 	return err;
 }
@@ -579,6 +593,8 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice)
 		destroy_stream(dice, AMDTP_IN_STREAM, i);
 		destroy_stream(dice, AMDTP_OUT_STREAM, i);
 	}
+
+	amdtp_domain_destroy(&dice->domain);
 }
 
 void snd_dice_stream_update_duplex(struct snd_dice *dice)
@@ -596,6 +612,8 @@ void snd_dice_stream_update_duplex(struct snd_dice *dice)
 	dice->global_enabled = false;
 
 	if (get_register_params(dice, &tx_params, &rx_params) == 0) {
+		amdtp_domain_stop(&dice->domain);
+
 		stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
 		stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
 	}
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index c6304e5e9fc4..fa6d74303f54 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -112,6 +112,8 @@ struct snd_dice {
 	bool global_enabled;
 	struct completion clock_accepted;
 	unsigned int substreams_counter;
+
+	struct amdtp_domain domain;
 };
 
 enum snd_dice_addr_type {
-- 
cgit v1.2.3-59-g8ed1b


From 9a08067ec318cbeaf0caa2d104cf677e723e02a3 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:34 +0900
Subject: ALSA: firewire-digi00x: support AMDTP domain

This commit adds AMDTP domain support for ALSA firewire-digi00x driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/digi00x/digi00x-stream.c | 55 +++++++++++++++++++--------------
 sound/firewire/digi00x/digi00x.h        |  2 ++
 2 files changed, 33 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c
index cff193f00a97..d6a92460060f 100644
--- a/sound/firewire/digi00x/digi00x-stream.c
+++ b/sound/firewire/digi00x/digi00x-stream.c
@@ -126,9 +126,6 @@ static void finish_session(struct snd_dg00x *dg00x)
 {
 	__be32 data;
 
-	amdtp_stream_stop(&dg00x->tx_stream);
-	amdtp_stream_stop(&dg00x->rx_stream);
-
 	data = cpu_to_be32(0x00000003);
 	snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
 			   DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET,
@@ -265,13 +262,23 @@ int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
 	if (err < 0)
 		destroy_stream(dg00x, &dg00x->rx_stream);
 
+	err = amdtp_domain_init(&dg00x->domain);
+	if (err < 0) {
+		destroy_stream(dg00x, &dg00x->rx_stream);
+		destroy_stream(dg00x, &dg00x->tx_stream);
+	}
+
 	return err;
 }
 
-// This function should be called before starting streams or after stopping
-// streams.
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
 void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x)
 {
+	amdtp_domain_destroy(&dg00x->domain);
+
 	destroy_stream(dg00x, &dg00x->rx_stream);
 	destroy_stream(dg00x, &dg00x->tx_stream);
 }
@@ -288,6 +295,8 @@ int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate)
 		rate = curr_rate;
 
 	if (dg00x->substreams_counter == 0 || curr_rate != rate) {
+		amdtp_domain_stop(&dg00x->domain);
+
 		finish_session(dg00x);
 
 		fw_iso_resources_free(&dg00x->tx_resources);
@@ -320,8 +329,10 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x)
 		return 0;
 
 	if (amdtp_streaming_error(&dg00x->tx_stream) ||
-	    amdtp_streaming_error(&dg00x->rx_stream))
+	    amdtp_streaming_error(&dg00x->rx_stream)) {
+		amdtp_domain_stop(&dg00x->domain);
 		finish_session(dg00x);
+	}
 
 	if (generation != fw_parent_device(dg00x->unit)->card->generation) {
 		err = fw_iso_resources_update(&dg00x->tx_resources);
@@ -338,36 +349,30 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x)
 	 * which source of clock is used.
 	 */
 	if (!amdtp_stream_running(&dg00x->rx_stream)) {
+		int spd = fw_parent_device(dg00x->unit)->max_speed;
+
 		err = begin_session(dg00x);
 		if (err < 0)
 			goto error;
 
-		err = amdtp_stream_start(&dg00x->rx_stream,
-				dg00x->rx_resources.channel,
-				fw_parent_device(dg00x->unit)->max_speed);
+		err = amdtp_domain_add_stream(&dg00x->domain, &dg00x->rx_stream,
+					      dg00x->rx_resources.channel, spd);
 		if (err < 0)
 			goto error;
 
-		if (!amdtp_stream_wait_callback(&dg00x->rx_stream,
-					      CALLBACK_TIMEOUT)) {
-			err = -ETIMEDOUT;
+		err = amdtp_domain_add_stream(&dg00x->domain, &dg00x->tx_stream,
+					      dg00x->tx_resources.channel, spd);
+		if (err < 0)
 			goto error;
-		}
-	}
 
-	/*
-	 * The value of SYT field in transmitted packets is always 0x0000. Thus,
-	 * duplex streams with timestamp synchronization cannot be built.
-	 */
-	if (!amdtp_stream_running(&dg00x->tx_stream)) {
-		err = amdtp_stream_start(&dg00x->tx_stream,
-				dg00x->tx_resources.channel,
-				fw_parent_device(dg00x->unit)->max_speed);
+		err = amdtp_domain_start(&dg00x->domain);
 		if (err < 0)
 			goto error;
 
-		if (!amdtp_stream_wait_callback(&dg00x->tx_stream,
-					      CALLBACK_TIMEOUT)) {
+		if (!amdtp_stream_wait_callback(&dg00x->rx_stream,
+						CALLBACK_TIMEOUT) ||
+		    !amdtp_stream_wait_callback(&dg00x->tx_stream,
+						CALLBACK_TIMEOUT)) {
 			err = -ETIMEDOUT;
 			goto error;
 		}
@@ -375,6 +380,7 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x)
 
 	return 0;
 error:
+	amdtp_domain_stop(&dg00x->domain);
 	finish_session(dg00x);
 
 	return err;
@@ -383,6 +389,7 @@ error:
 void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x)
 {
 	if (dg00x->substreams_counter == 0) {
+		amdtp_domain_stop(&dg00x->domain);
 		finish_session(dg00x);
 
 		fw_iso_resources_free(&dg00x->tx_resources);
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
index 0994d191ccda..8041c65f2736 100644
--- a/sound/firewire/digi00x/digi00x.h
+++ b/sound/firewire/digi00x/digi00x.h
@@ -59,6 +59,8 @@ struct snd_dg00x {
 
 	/* Console models have additional MIDI ports for control surface. */
 	bool is_console;
+
+	struct amdtp_domain domain;
 };
 
 #define DG00X_ADDR_BASE		0xffffe0000000ull
-- 
cgit v1.2.3-59-g8ed1b


From c281d46a51e31703183aa66ed9315446a0a10953 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:35 +0900
Subject: ALSA: firewire-tascam: support AMDTP domain

This commit adds AMDTP domain support for ALSA firewire-tascam driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/tascam/tascam-stream.c | 54 +++++++++++++++++++++--------------
 sound/firewire/tascam/tascam.h        |  2 ++
 2 files changed, 34 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
index d5e77036e0ee..9e2dc2fe3271 100644
--- a/sound/firewire/tascam/tascam-stream.c
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -180,9 +180,6 @@ static void finish_session(struct snd_tscm *tscm)
 {
 	__be32 reg;
 
-	amdtp_stream_stop(&tscm->rx_stream);
-	amdtp_stream_stop(&tscm->tx_stream);
-
 	reg = 0;
 	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
 			   TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
@@ -339,8 +336,16 @@ int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
 		return err;
 
 	err = init_stream(tscm, &tscm->rx_stream);
-	if (err < 0)
+	if (err < 0) {
+		destroy_stream(tscm, &tscm->tx_stream);
+		return err;
+	}
+
+	err = amdtp_domain_init(&tscm->domain);
+	if (err < 0) {
 		destroy_stream(tscm, &tscm->tx_stream);
+		destroy_stream(tscm, &tscm->rx_stream);
+	}
 
 	return err;
 }
@@ -348,17 +353,18 @@ int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
 // At bus reset, streaming is stopped and some registers are clear.
 void snd_tscm_stream_update_duplex(struct snd_tscm *tscm)
 {
-	amdtp_stream_pcm_abort(&tscm->tx_stream);
-	amdtp_stream_stop(&tscm->tx_stream);
+	amdtp_domain_stop(&tscm->domain);
 
+	amdtp_stream_pcm_abort(&tscm->tx_stream);
 	amdtp_stream_pcm_abort(&tscm->rx_stream);
-	amdtp_stream_stop(&tscm->rx_stream);
 }
 
 // This function should be called before starting streams or after stopping
 // streams.
 void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm)
 {
+	amdtp_domain_destroy(&tscm->domain);
+
 	destroy_stream(tscm, &tscm->rx_stream);
 	destroy_stream(tscm, &tscm->tx_stream);
 }
@@ -373,6 +379,8 @@ int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate)
 		return err;
 
 	if (tscm->substreams_counter == 0 || rate != curr_rate) {
+		amdtp_domain_stop(&tscm->domain);
+
 		finish_session(tscm);
 
 		fw_iso_resources_free(&tscm->tx_resources);
@@ -405,8 +413,10 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 		return 0;
 
 	if (amdtp_streaming_error(&tscm->rx_stream) ||
-	    amdtp_streaming_error(&tscm->tx_stream))
+	    amdtp_streaming_error(&tscm->tx_stream)) {
+		amdtp_domain_stop(&tscm->domain);
 		finish_session(tscm);
+	}
 
 	if (generation != fw_parent_device(tscm->unit)->card->generation) {
 		err = fw_iso_resources_update(&tscm->tx_resources);
@@ -419,6 +429,8 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 	}
 
 	if (!amdtp_stream_running(&tscm->rx_stream)) {
+		int spd = fw_parent_device(tscm->unit)->max_speed;
+
 		err = set_stream_formats(tscm, rate);
 		if (err < 0)
 			goto error;
@@ -427,27 +439,23 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 		if (err < 0)
 			goto error;
 
-		err = amdtp_stream_start(&tscm->rx_stream,
-				tscm->rx_resources.channel,
-				fw_parent_device(tscm->unit)->max_speed);
+		err = amdtp_domain_add_stream(&tscm->domain, &tscm->rx_stream,
+					      tscm->rx_resources.channel, spd);
 		if (err < 0)
 			goto error;
 
-		if (!amdtp_stream_wait_callback(&tscm->rx_stream,
-						CALLBACK_TIMEOUT)) {
-			err = -ETIMEDOUT;
+		err = amdtp_domain_add_stream(&tscm->domain, &tscm->tx_stream,
+					      tscm->tx_resources.channel, spd);
+		if (err < 0)
 			goto error;
-		}
-	}
 
-	if (!amdtp_stream_running(&tscm->tx_stream)) {
-		err = amdtp_stream_start(&tscm->tx_stream,
-				tscm->tx_resources.channel,
-				fw_parent_device(tscm->unit)->max_speed);
+		err = amdtp_domain_start(&tscm->domain);
 		if (err < 0)
-			goto error;
+			return err;
 
-		if (!amdtp_stream_wait_callback(&tscm->tx_stream,
+		if (!amdtp_stream_wait_callback(&tscm->rx_stream,
+						CALLBACK_TIMEOUT) ||
+		    !amdtp_stream_wait_callback(&tscm->tx_stream,
 						CALLBACK_TIMEOUT)) {
 			err = -ETIMEDOUT;
 			goto error;
@@ -456,6 +464,7 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 
 	return 0;
 error:
+	amdtp_domain_stop(&tscm->domain);
 	finish_session(tscm);
 
 	return err;
@@ -464,6 +473,7 @@ error:
 void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm)
 {
 	if (tscm->substreams_counter == 0) {
+		amdtp_domain_stop(&tscm->domain);
 		finish_session(tscm);
 
 		fw_iso_resources_free(&tscm->tx_resources);
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
index 734e5bb9c3da..64a2e4d2bbfe 100644
--- a/sound/firewire/tascam/tascam.h
+++ b/sound/firewire/tascam/tascam.h
@@ -97,6 +97,8 @@ struct snd_tscm {
 	struct snd_firewire_tascam_change queue[SND_TSCM_QUEUE_COUNT];
 	unsigned int pull_pos;
 	unsigned int push_pos;
+
+	struct amdtp_domain domain;
 };
 
 #define TSCM_ADDR_BASE			0xffff00000000ull
-- 
cgit v1.2.3-59-g8ed1b


From ccc6c1b0e604144d4d068eb676b444397cecf1b3 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:36 +0900
Subject: ALSA: firewire-motu: support AMDTP domain

This commit adds AMDTP domain support for ALSA firewire-motu driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/motu/motu-stream.c | 83 +++++++++++++++++++--------------------
 sound/firewire/motu/motu.h        |  2 +
 2 files changed, 42 insertions(+), 43 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c
index cc9f34426a47..813e38e6a86e 100644
--- a/sound/firewire/motu/motu-stream.c
+++ b/sound/firewire/motu/motu-stream.c
@@ -92,9 +92,6 @@ static void finish_session(struct snd_motu *motu)
 	if (err < 0)
 		return;
 
-	amdtp_stream_stop(&motu->tx_stream);
-	amdtp_stream_stop(&motu->rx_stream);
-
 	err = snd_motu_transaction_read(motu, ISOC_COMM_CONTROL_OFFSET, &reg,
 					sizeof(reg));
 	if (err < 0)
@@ -109,27 +106,6 @@ static void finish_session(struct snd_motu *motu)
 				   sizeof(reg));
 }
 
-static int start_isoc_ctx(struct snd_motu *motu, struct amdtp_stream *stream)
-{
-	struct fw_iso_resources *resources;
-	int err;
-
-	if (stream == &motu->rx_stream)
-		resources = &motu->rx_resources;
-	else
-		resources = &motu->tx_resources;
-
-	err = amdtp_stream_start(stream, resources->channel,
-				 fw_parent_device(motu->unit)->max_speed);
-	if (err < 0)
-		return err;
-
-	if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT))
-		return -ETIMEDOUT;
-
-	return 0;
-}
-
 int snd_motu_stream_cache_packet_formats(struct snd_motu *motu)
 {
 	int err;
@@ -169,6 +145,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate)
 		rate = curr_rate;
 
 	if (motu->substreams_counter == 0 || curr_rate != rate) {
+		amdtp_domain_stop(&motu->domain);
 		finish_session(motu);
 
 		fw_iso_resources_free(&motu->tx_resources);
@@ -234,8 +211,10 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu)
 		return 0;
 
 	if (amdtp_streaming_error(&motu->rx_stream) ||
-	    amdtp_streaming_error(&motu->tx_stream))
+	    amdtp_streaming_error(&motu->tx_stream)) {
+		amdtp_domain_stop(&motu->domain);
 		finish_session(motu);
+	}
 
 	if (generation != fw_parent_device(motu->unit)->card->generation) {
 		err = fw_iso_resources_update(&motu->rx_resources);
@@ -248,6 +227,8 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu)
 	}
 
 	if (!amdtp_stream_running(&motu->rx_stream)) {
+		int spd = fw_parent_device(motu->unit)->max_speed;
+
 		err = ensure_packet_formats(motu);
 		if (err < 0)
 			return err;
@@ -259,26 +240,32 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu)
 			goto stop_streams;
 		}
 
-		err = start_isoc_ctx(motu, &motu->rx_stream);
-		if (err < 0) {
-			dev_err(&motu->unit->device,
-				"fail to start IT context: %d\n", err);
+		err = amdtp_domain_add_stream(&motu->domain, &motu->tx_stream,
+					      motu->tx_resources.channel, spd);
+		if (err < 0)
 			goto stop_streams;
-		}
 
-		err = motu->spec->protocol->switch_fetching_mode(motu, true);
-		if (err < 0) {
-			dev_err(&motu->unit->device,
-				"fail to enable frame fetching: %d\n", err);
+		err = amdtp_domain_add_stream(&motu->domain, &motu->rx_stream,
+					      motu->rx_resources.channel, spd);
+		if (err < 0)
+			goto stop_streams;
+
+		err = amdtp_domain_start(&motu->domain);
+		if (err < 0)
+			goto stop_streams;
+
+		if (!amdtp_stream_wait_callback(&motu->tx_stream,
+						CALLBACK_TIMEOUT) ||
+		    !amdtp_stream_wait_callback(&motu->rx_stream,
+						CALLBACK_TIMEOUT)) {
+			err = -ETIMEDOUT;
 			goto stop_streams;
 		}
-	}
 
-	if (!amdtp_stream_running(&motu->tx_stream)) {
-		err = start_isoc_ctx(motu, &motu->tx_stream);
+		err = motu->spec->protocol->switch_fetching_mode(motu, true);
 		if (err < 0) {
 			dev_err(&motu->unit->device,
-				"fail to start IR context: %d", err);
+				"fail to enable frame fetching: %d\n", err);
 			goto stop_streams;
 		}
 	}
@@ -286,6 +273,7 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu)
 	return 0;
 
 stop_streams:
+	amdtp_domain_stop(&motu->domain);
 	finish_session(motu);
 	return err;
 }
@@ -293,6 +281,7 @@ stop_streams:
 void snd_motu_stream_stop_duplex(struct snd_motu *motu)
 {
 	if (motu->substreams_counter == 0) {
+		amdtp_domain_stop(&motu->domain);
 		finish_session(motu);
 
 		fw_iso_resources_free(&motu->tx_resources);
@@ -344,18 +333,26 @@ int snd_motu_stream_init_duplex(struct snd_motu *motu)
 		return err;
 
 	err = init_stream(motu, &motu->rx_stream);
-	if (err < 0)
+	if (err < 0) {
 		destroy_stream(motu, &motu->tx_stream);
+		return err;
+	}
+
+	err = amdtp_domain_init(&motu->domain);
+	if (err < 0) {
+		destroy_stream(motu, &motu->tx_stream);
+		destroy_stream(motu, &motu->rx_stream);
+	}
 
 	return err;
 }
 
-/*
- * This function should be called before starting streams or after stopping
- * streams.
- */
+// This function should be called before starting streams or after stopping
+// streams.
 void snd_motu_stream_destroy_duplex(struct snd_motu *motu)
 {
+	amdtp_domain_destroy(&motu->domain);
+
 	destroy_stream(motu, &motu->rx_stream);
 	destroy_stream(motu, &motu->tx_stream);
 
diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h
index 09d1451d7de4..350ee2c16f4a 100644
--- a/sound/firewire/motu/motu.h
+++ b/sound/firewire/motu/motu.h
@@ -69,6 +69,8 @@ struct snd_motu {
 	int dev_lock_count;
 	bool dev_lock_changed;
 	wait_queue_head_t hwdep_wait;
+
+	struct amdtp_domain domain;
 };
 
 enum snd_motu_spec_flags {
-- 
cgit v1.2.3-59-g8ed1b


From c9a9ce899cd1798f054f264043bddc021b47cb1e Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:37 +0900
Subject: ALSA: fireface: support AMDTP domain

This commit adds AMDTP domain support for ALSA fireface driver.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/fireface/ff-stream.c | 57 +++++++++++++++++++++----------------
 sound/firewire/fireface/ff.h        |  2 ++
 2 files changed, 35 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c
index e4710204f481..e8e6f9fd6433 100644
--- a/sound/firewire/fireface/ff-stream.c
+++ b/sound/firewire/fireface/ff-stream.c
@@ -32,9 +32,6 @@ int snd_ff_stream_get_multiplier_mode(enum cip_sfc sfc,
 
 static inline void finish_session(struct snd_ff *ff)
 {
-	amdtp_stream_stop(&ff->tx_stream);
-	amdtp_stream_stop(&ff->rx_stream);
-
 	ff->spec->protocol->finish_session(ff);
 	ff->spec->protocol->switch_fetching_mode(ff, false);
 }
@@ -83,8 +80,16 @@ int snd_ff_stream_init_duplex(struct snd_ff *ff)
 		return err;
 
 	err = init_stream(ff, &ff->tx_stream);
-	if (err < 0)
+	if (err < 0) {
 		destroy_stream(ff, &ff->rx_stream);
+		return err;
+	}
+
+	err = amdtp_domain_init(&ff->domain);
+	if (err < 0) {
+		destroy_stream(ff, &ff->rx_stream);
+		destroy_stream(ff, &ff->tx_stream);
+	}
 
 	return err;
 }
@@ -95,6 +100,8 @@ int snd_ff_stream_init_duplex(struct snd_ff *ff)
  */
 void snd_ff_stream_destroy_duplex(struct snd_ff *ff)
 {
+	amdtp_domain_destroy(&ff->domain);
+
 	destroy_stream(ff, &ff->rx_stream);
 	destroy_stream(ff, &ff->tx_stream);
 }
@@ -113,6 +120,7 @@ int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate)
 		enum snd_ff_stream_mode mode;
 		int i;
 
+		amdtp_domain_stop(&ff->domain);
 		finish_session(ff);
 
 		fw_iso_resources_free(&ff->tx_resources);
@@ -155,51 +163,52 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate)
 		return 0;
 
 	if (amdtp_streaming_error(&ff->tx_stream) ||
-	    amdtp_streaming_error(&ff->rx_stream))
+	    amdtp_streaming_error(&ff->rx_stream)) {
+		amdtp_domain_stop(&ff->domain);
 		finish_session(ff);
+	}
 
 	/*
 	 * Regardless of current source of clock signal, drivers transfer some
 	 * packets. Then, the device transfers packets.
 	 */
 	if (!amdtp_stream_running(&ff->rx_stream)) {
+		int spd = fw_parent_device(ff->unit)->max_speed;
+
 		err = ff->spec->protocol->begin_session(ff, rate);
 		if (err < 0)
 			goto error;
 
-		err = amdtp_stream_start(&ff->rx_stream,
-					 ff->rx_resources.channel,
-					 fw_parent_device(ff->unit)->max_speed);
+		err = amdtp_domain_add_stream(&ff->domain, &ff->rx_stream,
+					      ff->rx_resources.channel, spd);
 		if (err < 0)
 			goto error;
 
-		if (!amdtp_stream_wait_callback(&ff->rx_stream,
-						CALLBACK_TIMEOUT_MS)) {
-			err = -ETIMEDOUT;
-			goto error;
-		}
-
-		err = ff->spec->protocol->switch_fetching_mode(ff, true);
+		err = amdtp_domain_add_stream(&ff->domain, &ff->tx_stream,
+					      ff->tx_resources.channel, spd);
 		if (err < 0)
 			goto error;
-	}
 
-	if (!amdtp_stream_running(&ff->tx_stream)) {
-		err = amdtp_stream_start(&ff->tx_stream,
-					 ff->tx_resources.channel,
-					 fw_parent_device(ff->unit)->max_speed);
+		err = amdtp_domain_start(&ff->domain);
 		if (err < 0)
 			goto error;
 
-		if (!amdtp_stream_wait_callback(&ff->tx_stream,
+		if (!amdtp_stream_wait_callback(&ff->rx_stream,
+						CALLBACK_TIMEOUT_MS) ||
+		    !amdtp_stream_wait_callback(&ff->tx_stream,
 						CALLBACK_TIMEOUT_MS)) {
 			err = -ETIMEDOUT;
 			goto error;
 		}
+
+		err = ff->spec->protocol->switch_fetching_mode(ff, true);
+		if (err < 0)
+			goto error;
 	}
 
 	return 0;
 error:
+	amdtp_domain_stop(&ff->domain);
 	finish_session(ff);
 
 	return err;
@@ -208,6 +217,7 @@ error:
 void snd_ff_stream_stop_duplex(struct snd_ff *ff)
 {
 	if (ff->substreams_counter == 0) {
+		amdtp_domain_stop(&ff->domain);
 		finish_session(ff);
 
 		fw_iso_resources_free(&ff->tx_resources);
@@ -217,12 +227,11 @@ void snd_ff_stream_stop_duplex(struct snd_ff *ff)
 
 void snd_ff_stream_update_duplex(struct snd_ff *ff)
 {
+	amdtp_domain_stop(&ff->domain);
+
 	// The device discontinue to transfer packets.
 	amdtp_stream_pcm_abort(&ff->tx_stream);
-	amdtp_stream_stop(&ff->tx_stream);
-
 	amdtp_stream_pcm_abort(&ff->rx_stream);
-	amdtp_stream_stop(&ff->rx_stream);
 }
 
 void snd_ff_stream_lock_changed(struct snd_ff *ff)
diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h
index 36dd0c75b9f7..b4c22ca6079e 100644
--- a/sound/firewire/fireface/ff.h
+++ b/sound/firewire/fireface/ff.h
@@ -91,6 +91,8 @@ struct snd_ff {
 	int dev_lock_count;
 	bool dev_lock_changed;
 	wait_queue_head_t hwdep_wait;
+
+	struct amdtp_domain domain;
 };
 
 enum snd_ff_clock_src {
-- 
cgit v1.2.3-59-g8ed1b


From 74f94e41825549638a5b24410c1f62d5ea57134e Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Sun, 4 Aug 2019 15:21:38 +0900
Subject: ALSA: firewire-lib: localize kernel APIs to start/stop each AMDTP
 stream

As a result to support AMDTP domain, no drivers call kernel APIs to
start/stop each AMDTP stream. This commit localize these APIs.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 6 ++----
 sound/firewire/amdtp-stream.h | 2 --
 2 files changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 158d210caea7..1a92855c7647 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -894,7 +894,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context,
  * amdtp_stream_set_parameters() and it must be started before any PCM or MIDI
  * device can be started.
  */
-int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed)
+static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed)
 {
 	static const struct {
 		unsigned int data_block;
@@ -1027,7 +1027,6 @@ err_unlock:
 
 	return err;
 }
-EXPORT_SYMBOL(amdtp_stream_start);
 
 /**
  * amdtp_stream_pcm_pointer - get the PCM buffer position
@@ -1098,7 +1097,7 @@ EXPORT_SYMBOL(amdtp_stream_update);
  * All PCM and MIDI devices of the stream must be stopped before the stream
  * itself can be stopped.
  */
-void amdtp_stream_stop(struct amdtp_stream *s)
+static void amdtp_stream_stop(struct amdtp_stream *s)
 {
 	mutex_lock(&s->mutex);
 
@@ -1118,7 +1117,6 @@ void amdtp_stream_stop(struct amdtp_stream *s)
 
 	mutex_unlock(&s->mutex);
 }
-EXPORT_SYMBOL(amdtp_stream_stop);
 
 /**
  * amdtp_stream_pcm_abort - abort the running PCM device
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 15d471660a43..bbbca964b9b4 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -188,9 +188,7 @@ int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
 				unsigned int data_block_quadlets);
 unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s);
 
-int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed);
 void amdtp_stream_update(struct amdtp_stream *s);
-void amdtp_stream_stop(struct amdtp_stream *s);
 
 int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 					struct snd_pcm_runtime *runtime);
-- 
cgit v1.2.3-59-g8ed1b


From c1ae5e7f057beb1daa72831ab22ec06e71669af7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 27 Jul 2019 10:55:49 +0200
Subject: ALSA: usb-audio: Unify audioformat release code

There are many open code for releasing audioformat object.
Provide a unified helper and call it from the all places.

Only a cleanup, no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/stream.c | 31 ++++++++++++++-----------------
 1 file changed, 14 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index e852c7fd6109..963d425004f8 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -28,6 +28,14 @@
 #include "power.h"
 #include "media.h"
 
+static void audioformat_free(struct audioformat *fp)
+{
+	list_del(&fp->list); /* unlink for avoiding double-free */
+	kfree(fp->rate_table);
+	kfree(fp->chmap);
+	kfree(fp);
+}
+
 /*
  * free a substream
  */
@@ -37,11 +45,8 @@ static void free_substream(struct snd_usb_substream *subs)
 
 	if (!subs->num_formats)
 		return; /* not initialized */
-	list_for_each_entry_safe(fp, n, &subs->fmt_list, list) {
-		kfree(fp->rate_table);
-		kfree(fp->chmap);
-		kfree(fp);
-	}
+	list_for_each_entry_safe(fp, n, &subs->fmt_list, list)
+		audioformat_free(fp);
 	kfree(subs->rate_list.list);
 	kfree(subs->str_pd);
 	snd_media_stream_delete(subs);
@@ -832,8 +837,7 @@ found_clock:
 	/* ok, let's parse further... */
 	if (snd_usb_parse_audio_format(chip, fp, format,
 					fmt, stream) < 0) {
-		kfree(fp->rate_table);
-		kfree(fp);
+		audioformat_free(fp);
 		return NULL;
 	}
 
@@ -1043,9 +1047,7 @@ found_clock:
 
 		pd = kzalloc(sizeof(*pd), GFP_KERNEL);
 		if (!pd) {
-			kfree(fp->chmap);
-			kfree(fp->rate_table);
-			kfree(fp);
+			audioformat_free(fp);
 			return NULL;
 		}
 		pd->pd_id = (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
@@ -1064,9 +1066,7 @@ found_clock:
 		/* ok, let's parse further... */
 		if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) {
 			kfree(pd);
-			kfree(fp->chmap);
-			kfree(fp->rate_table);
-			kfree(fp);
+			audioformat_free(fp);
 			return NULL;
 		}
 	}
@@ -1185,11 +1185,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 			err = snd_usb_add_audio_stream(chip, stream, fp);
 
 		if (err < 0) {
-			list_del(&fp->list); /* unlink for avoiding double-free */
+			audioformat_free(fp);
 			kfree(pd);
-			kfree(fp->rate_table);
-			kfree(fp->chmap);
-			kfree(fp);
 			return err;
 		}
 		/* try to set the interface... */
-- 
cgit v1.2.3-59-g8ed1b


From f7f530181461aaf6afa0b3f0be4d239dea410896 Mon Sep 17 00:00:00 2001
From: Alexander Tsoy <alexander@tsoy.me>
Date: Fri, 11 Aug 2017 02:36:14 +0300
Subject: ALSA: usb-audio: fix PCM device order

Some cards have alternate setting with non-PCM format as the first
altsetting in the interface descriptors. This confuses userspace, since
alsa-lib uses device 0 by default. So lets parse interfaces in two steps:
 1. Parse altsettings with PCM formats.
 2. Parse altsettings with non-PCM formats.

This fixes at least following cards:
 - Audinst HUD-mx2
 - Audinst HUD-mini

[ Adapted to 5.3 kernel by tiwai ]

Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/stream.c | 34 +++++++++++++++++++++++++++++++++-
 1 file changed, 33 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 963d425004f8..fc3e9fcfbc38 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -1077,7 +1077,9 @@ found_clock:
 	return fp;
 }
 
-int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
+static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
+					   int iface_no,
+					   bool *has_non_pcm, bool non_pcm)
 {
 	struct usb_device *dev;
 	struct usb_interface *iface;
@@ -1178,6 +1180,16 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 		else if (IS_ERR(fp))
 			return PTR_ERR(fp);
 
+		if (fp->fmt_type != UAC_FORMAT_TYPE_I)
+			*has_non_pcm = true;
+		if ((fp->fmt_type == UAC_FORMAT_TYPE_I) == non_pcm) {
+			audioformat_free(fp);
+			kfree(pd);
+			fp = NULL;
+			pd = NULL;
+			continue;
+		}
+
 		dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
 		if (protocol == UAC_VERSION_3)
 			err = snd_usb_add_audio_stream_v3(chip, stream, fp, pd);
@@ -1197,3 +1209,23 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 	return 0;
 }
 
+int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
+{
+	int err;
+	bool has_non_pcm = false;
+
+	/* parse PCM formats */
+	err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, false);
+	if (err < 0)
+		return err;
+
+	if (has_non_pcm) {
+		/* parse non-PCM formats */
+		err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, true);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
-- 
cgit v1.2.3-59-g8ed1b


From 1b34121d9f26d272b0b2334209af6b6fc82d4bf1 Mon Sep 17 00:00:00 2001
From: Ard van Breemen <ard@kwaak.net>
Date: Fri, 2 Aug 2019 13:52:14 +0200
Subject: ALSA: usb-audio: Skip bSynchAddress endpoint check if it is invalid

The Linux kernel assumes that get_endpoint(alts,0) and
get_endpoint(alts,1) are eachothers feedback endpoints.
To reassure that validity it will test bsynchaddress to comply with that
assumption. But if the bsyncaddress is 0 (invalid), it will flag that as
a wrong assumption and return an error.
Fix: Skip the test if bSynchAddress is 0.
Note: those with a valid bSynchAddress should have a code quirck added.

Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 75b96929f76c..51a8faf25ed8 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -456,6 +456,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 	}
 	ep = get_endpoint(alts, 1)->bEndpointAddress;
 	if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	    get_endpoint(alts, 0)->bSynchAddress != 0 &&
 	    ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
 	     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
 		dev_err(&dev->dev,
-- 
cgit v1.2.3-59-g8ed1b


From 118b2806a07605d6e3c880e35a5c472bfda7582f Mon Sep 17 00:00:00 2001
From: Ard van Breemen <ard@kwaak.net>
Date: Fri, 2 Aug 2019 13:52:15 +0200
Subject: ALSA: usb-audio: Add Pioneer DDJ-SX3 PCM quirck

The Pioneer DDJ-SX3 is a plain 12 32bit channel out and 10 channel in
PCM/midi controller. The PCM part is "vendor specific".
It needs the "ignore invalid bsynchaddress" patch as it uses 0 for that.

Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 57 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e918ce346027..70c338f3ae24 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3534,5 +3534,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		}
 	}
 },
+{
+	/*
+	 * PIONEER DJ DDJ-SX3
+	 * PCM is 12 channels out, 10 channels in @ 44.1 fixed
+	 * interface 0, vendor class alt setting 1 for endpoints 5 and 0x86
+	 * The feedback for the output is the input.
+	 */
+	USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0023),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S32_LE,
+					.channels = 12,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x05,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						   USB_ENDPOINT_SYNC_ASYNC,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S32_LE,
+					.channels = 10,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.endpoint = 0x86,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC|
+						 USB_ENDPOINT_SYNC_ASYNC|
+						 USB_ENDPOINT_USAGE_IMPLICIT_FB,
+					.rates = SNDRV_PCM_RATE_44100,
+					.rate_min = 44100,
+					.rate_max = 44100,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 44100 }
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 #undef USB_DEVICE_VENDOR_SPEC
-- 
cgit v1.2.3-59-g8ed1b


From 06e8f5c842f2dbb232897ba967ea7b422745c271 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 6 Aug 2019 12:45:38 +0900
Subject: ASoC: rsnd: don't call clk_get_rate() under atomic context

ADG is using clk_get_rate() under atomic context, thus, we might
have scheduling issue.
To avoid this issue, we need to get/keep clk rate under
non atomic context.

We need to handle ADG as special device at Renesas Sound driver.
From SW point of view, we want to impletent it as
rsnd_mod_ops :: prepare, but it makes code just complicate.

To avoid complicated code/patch, this patch adds new clk_rate[] array,
and keep clk IN rate when rsnd_adg_clk_enable() was called.

Reported-by: Leon Kong <Leon.KONG@cn.bosch.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Leon Kong <Leon.KONG@cn.bosch.com>
Link: https://lore.kernel.org/r/87v9vb0xkp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sh/rcar/adg.c | 21 +++++++++++++++------
 1 file changed, 15 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index fce4e050a9b7..b9aacf3d3b29 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -30,6 +30,7 @@ struct rsnd_adg {
 	struct clk *clkout[CLKOUTMAX];
 	struct clk_onecell_data onecell;
 	struct rsnd_mod mod;
+	int clk_rate[CLKMAX];
 	u32 flags;
 	u32 ckr;
 	u32 rbga;
@@ -114,9 +115,9 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv,
 	unsigned int val, en;
 	unsigned int min, diff;
 	unsigned int sel_rate[] = {
-		clk_get_rate(adg->clk[CLKA]),	/* 0000: CLKA */
-		clk_get_rate(adg->clk[CLKB]),	/* 0001: CLKB */
-		clk_get_rate(adg->clk[CLKC]),	/* 0010: CLKC */
+		adg->clk_rate[CLKA],	/* 0000: CLKA */
+		adg->clk_rate[CLKB],	/* 0001: CLKB */
+		adg->clk_rate[CLKC],	/* 0010: CLKC */
 		adg->rbga_rate_for_441khz,	/* 0011: RBGA */
 		adg->rbgb_rate_for_48khz,	/* 0100: RBGB */
 	};
@@ -302,7 +303,7 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate)
 	 * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI.
 	 */
 	for_each_rsnd_clk(clk, adg, i) {
-		if (rate == clk_get_rate(clk))
+		if (rate == adg->clk_rate[i])
 			return sel_table[i];
 	}
 
@@ -369,10 +370,18 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 
 	for_each_rsnd_clk(clk, adg, i) {
 		ret = 0;
-		if (enable)
+		if (enable) {
 			ret = clk_prepare_enable(clk);
-		else
+
+			/*
+			 * We shouldn't use clk_get_rate() under
+			 * atomic context. Let's keep it when
+			 * rsnd_adg_clk_enable() was called
+			 */
+			adg->clk_rate[i] = clk_get_rate(adg->clk[i]);
+		} else {
 			clk_disable_unprepare(clk);
+		}
 
 		if (ret < 0)
 			dev_warn(dev, "can't use clk %d\n", i);
-- 
cgit v1.2.3-59-g8ed1b


From cc352735d49c9c25fd711795bee35d3d001ddffa Mon Sep 17 00:00:00 2001
From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Date: Tue, 6 Aug 2019 15:19:58 -0700
Subject: ASoC: SOF: Intel: hda: Initialize HDA controller after i915 init

On some platforms, sound card registration fails when a HDMI
monitor is not connected. This is caused by a recent commit
that switched the order in which the HDA controller and the
i915 are initialized. Initializing the i915 before initializing
the HDA controller fixes the problem.

Fixes: be1b577d01787c ("ASoC: SOF: Intel: hda: fix the hda init chip"
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20190806221958.19180-1-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda.c | 20 +++++++++++++-------
 1 file changed, 13 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 136f98bf5d7e..82f61c4e74bd 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -321,10 +321,23 @@ static int hda_init_caps(struct snd_sof_dev *sdev)
 	if (bus->ppcap)
 		dev_dbg(sdev->dev, "PP capability, will probe DSP later.\n");
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	/* init i915 and HDMI codecs */
+	ret = hda_codec_i915_init(sdev);
+	if (ret < 0) {
+		dev_err(sdev->dev, "error: init i915 and HDMI codec failed\n");
+		return ret;
+	}
+#endif
+
+	/* Init HDA controller after i915 init */
 	ret = hda_dsp_ctrl_init_chip(sdev, true);
 	if (ret < 0) {
 		dev_err(bus->dev, "error: init chip failed with ret: %d\n",
 			ret);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+		hda_codec_i915_exit(sdev);
+#endif
 		return ret;
 	}
 
@@ -332,13 +345,6 @@ static int hda_init_caps(struct snd_sof_dev *sdev)
 	if (bus->mlcap)
 		snd_hdac_ext_bus_get_ml_capabilities(bus);
 
-	/* init i915 and HDMI codecs */
-	ret = hda_codec_i915_init(sdev);
-	if (ret < 0) {
-		dev_err(sdev->dev, "error: no HDMI audio devices found\n");
-		return ret;
-	}
-
 	/* codec detection */
 	if (!bus->codec_mask) {
 		dev_info(bus->dev, "no hda codecs found!\n");
-- 
cgit v1.2.3-59-g8ed1b


From 3403b808a0eb6d25883b72a6c4d2a4a452f8a233 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Tue, 6 Aug 2019 17:14:35 +0800
Subject: ASoC: rt1011: Add R0 temperature and TDM1 ADC2DAT Swap control

- The user level application could set the R0 temperature after booting system.
  The degree Celsius of R0 temperature store in the non-volatile space
  when doing R0 calibration.
- TDM1 ADC2DAT Swap controls use to control TDM slot2/3 data

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20190806091435.14329-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1011.c | 9 ++++++++-
 sound/soc/codecs/rt1011.h | 1 +
 2 files changed, 9 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 0a6ff13d76e1..b25b4c1127ca 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -1029,6 +1029,8 @@ static const char * const rt1011_tdm_adc_swap_select[] = {
 
 static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum,	RT1011_TDM1_SET_3, 6,
 	rt1011_tdm_adc_swap_select);
+static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum,	RT1011_TDM1_SET_3, 4,
+	rt1011_tdm_adc_swap_select);
 
 static void rt1011_reset(struct regmap *regmap)
 {
@@ -1333,7 +1335,8 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = {
 	/* TDM1 Data Out Selection */
 	SOC_ENUM("TDM1 DOUT Source", rt1011_tdm1_adc1_dat_enum),
 	SOC_ENUM("TDM1 DOUT Location", rt1011_tdm1_adc1_loc_enum),
-	SOC_ENUM("TDM1 ADCDAT Swap Select", rt1011_tdm_adc1_1_enum),
+	SOC_ENUM("TDM1 ADC1DAT Swap Select", rt1011_tdm_adc1_1_enum),
+	SOC_ENUM("TDM1 ADC2DAT Swap Select", rt1011_tdm_adc2_1_enum),
 
 	/* Data Out Mode */
 	SOC_ENUM("I2S ADC DOUT Mode", rt1011_adc_dout_mode_enum),
@@ -1355,6 +1358,10 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = {
 	SOC_SINGLE_EXT("R0 Calibration", SND_SOC_NOPM, 0, 1, 0,
 		rt1011_r0_cali_get, rt1011_r0_cali_put),
 	RT1011_R0_LOAD("R0 Load Mode"),
+
+	/* R0 temperature */
+	SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP,
+		2, 255, 0),
 };
 
 static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h
index 98a38800c4df..5d431d7f1d16 100644
--- a/sound/soc/codecs/rt1011.h
+++ b/sound/soc/codecs/rt1011.h
@@ -227,6 +227,7 @@
 #define RT1011_STP_CALIB_RS_TEMP			0x152a
 #define RT1011_INIT_RECIPROCAL_REG_24_16				0x1538
 #define RT1011_INIT_RECIPROCAL_REG_15_0				0x1539
+#define RT1011_STP_INITIAL_RESISTANCE_TEMP				0x153c
 #define RT1011_STP_ALPHA_RECIPROCAL_MSB				0x153e
 #define RT1011_SPK_RESISTANCE_1				0x1544
 #define RT1011_SPK_RESISTANCE_2				0x1546
-- 
cgit v1.2.3-59-g8ed1b


From 0922c7a51c775b71a297598623debe9cb6ce3a08 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Tue, 6 Aug 2019 17:14:59 +0800
Subject: ASoC: rt1011: Add a flag for the R0 calibration test

The factory test needs to know whether the calibration completed.
This flag helps to confirm the calibration completed or not.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20190806091459.14382-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1011.c | 14 +++++++++++++-
 sound/soc/codecs/rt1011.h |  2 +-
 2 files changed, 14 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index b25b4c1127ca..638abcaf52b3 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -1225,7 +1225,10 @@ static int rt1011_bq_drc_info(struct snd_kcontrol *kcontrol,
 static int rt1011_r0_cali_get(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
-	ucontrol->value.integer.value[0] = 0;
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component);
+
+	ucontrol->value.integer.value[0] = rt1011->cali_done;
 
 	return 0;
 }
@@ -1239,6 +1242,7 @@ static int rt1011_r0_cali_put(struct snd_kcontrol *kcontrol,
 	if (!component->card->instantiated)
 		return 0;
 
+	rt1011->cali_done = 0;
 	if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF &&
 		ucontrol->value.integer.value[0])
 		rt1011_calibrate(rt1011, 1);
@@ -2135,6 +2139,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag)
 			r0_factor = ((format / r0[0] * 100) / 128)
 							- (r0_integer * 100);
 			rt1011->r0_reg = r0[0];
+			rt1011->cali_done = 1;
 			dev_info(dev,	"r0 resistance about %d.%02d ohm, reg=0x%X\n",
 				r0_integer, r0_factor, r0[0]);
 		}
@@ -2185,6 +2190,13 @@ static void rt1011_calibration_work(struct work_struct *work)
 
 	rt1011_calibrate(rt1011, 1);
 
+	/*
+	 * This flag should reset after booting.
+	 * The factory test will do calibration again and use this flag to check
+	 * whether the calibration completed
+	 */
+	rt1011->cali_done = 0;
+
 	/* initial */
 	rt1011_reg_init(component);
 }
diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h
index 5d431d7f1d16..2d65983f3d0f 100644
--- a/sound/soc/codecs/rt1011.h
+++ b/sound/soc/codecs/rt1011.h
@@ -666,7 +666,7 @@ struct rt1011_priv {
 	int pll_out;
 
 	int bq_drc_set;
-	unsigned int r0_reg;
+	unsigned int r0_reg, cali_done;
 	int recv_spk_mode;
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From bb67dd1878de5766c24fe7a4002d1a12c14da518 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Tue, 6 Aug 2019 12:06:03 -0500
Subject: ASoC: SOF: Intel: hda: fix MSI handling

The addition of a kernel module parameter to optionally disable MSI
had the side effect of permanently disabling it.

The return value of pci_alloc_irq_vectors() is the number of allocated
vectors or a negative number on error, so testing with the ! operator
is not quite right. It was one optimization too far.

Restore previous behavior to use MSI by default, unless the user
selects not to do so or the allocation of irq_vectors fails.

Fixes: 672ff5e3596ee ('ASoC: SOF: Intel: hda: add a parameter to disable MSI')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190806170603.10815-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 79cce20666b6..53813de7844e 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -537,7 +537,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev)
 	 * if it fails, use legacy interrupt mode
 	 * TODO: support msi multiple vectors
 	 */
-	if (hda_use_msi && !pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI)) {
+	if (hda_use_msi && pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI) > 0) {
 		dev_info(sdev->dev, "use msi interrupt mode\n");
 		hdev->irq = pci_irq_vector(pci, 0);
 		/* ipc irq number is the same of hda irq */
-- 
cgit v1.2.3-59-g8ed1b


From ad64bfbd09d7b7997a1b1510983350836bcd6ed9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:30:13 +0900
Subject: ASoC: soc-core: set component->debugfs_root NULL

To be more safety code, let's set NULL to component->debugfs_root
when it was cleanuped.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87muglahq0.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0f75dac4bb26..2f37c4e0974b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -171,7 +171,10 @@ static void soc_init_component_debugfs(struct snd_soc_component *component)
 
 static void soc_cleanup_component_debugfs(struct snd_soc_component *component)
 {
+	if (!component->debugfs_root)
+		return;
 	debugfs_remove_recursive(component->debugfs_root);
+	component->debugfs_root = NULL;
 }
 
 static int dai_list_show(struct seq_file *m, void *v)
-- 
cgit v1.2.3-59-g8ed1b


From 6634e3d6ea8cd92fce10d1d2da0c8e42ab84da32 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:30:31 +0900
Subject: ASoC: soc-core: add comment for for_each_xxx

soc-core has many for_each_xxx, but it is a little bit
difficult to know which list is relead to which for_each_xxx.
This patch adds missing comment for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lfw5ahpj.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2f37c4e0974b..5e6a48a0f79e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -380,6 +380,7 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd)
 static void soc_add_pcm_runtime(struct snd_soc_card *card,
 		struct snd_soc_pcm_runtime *rtd)
 {
+	/* see for_each_card_rtds */
 	list_add_tail(&rtd->list, &card->rtd_list);
 	rtd->num = card->num_rtd;
 	card->num_rtd++;
@@ -1149,6 +1150,7 @@ int snd_soc_add_dai_link(struct snd_soc_card *card,
 	if (dai_link->dobj.type && card->add_dai_link)
 		card->add_dai_link(card, dai_link);
 
+	/* see for_each_card_links */
 	list_add_tail(&dai_link->list, &card->dai_link_list);
 
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From 5b99a0aad08a3428cc3262ecee29a71c88c981c2 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:30:36 +0900
Subject: ASoC: soc-core: check return value of snd_soc_add_dai_link()

snd_soc_add_dai_link() might return error, we need to check it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k1bpahpd.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 5e6a48a0f79e..7345679d4903 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1964,8 +1964,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* add predefined DAI links to the list */
-	for_each_card_prelinks(card, i, dai_link)
-		snd_soc_add_dai_link(card, dai_link);
+	for_each_card_prelinks(card, i, dai_link) {
+		ret = snd_soc_add_dai_link(card, dai_link);
+		if (ret < 0)
+			goto probe_end;
+	}
 
 	/* card bind complete so register a sound card */
 	ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
-- 
cgit v1.2.3-59-g8ed1b


From 42849064500b24892e2abfe8679507dfffd585f1 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:30:41 +0900
Subject: ASoC: soc-core: don't use for_each_card_links_safe() at
 snd_soc_find_dai_link()

It doesn't removes list during loop at snd_soc_find_dai_link().
We don't need to use _safe loop.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87imr9ahp9.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7345679d4903..e176b972e4e6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -817,11 +817,11 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card,
 					       int id, const char *name,
 					       const char *stream_name)
 {
-	struct snd_soc_dai_link *link, *_link;
+	struct snd_soc_dai_link *link;
 
 	lockdep_assert_held(&client_mutex);
 
-	for_each_card_links_safe(card, link, _link) {
+	for_each_card_links(card, link) {
 		if (link->id != id)
 			continue;
 
-- 
cgit v1.2.3-59-g8ed1b


From 32d2c172fe88705058310635a732259c0a98e89b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:30:47 +0900
Subject: ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add()

snd_soc_rtdcom_add() is using both "rtdcom" and "new_rtdcom" as
variable name, but these are not used at same time.
Let's reuse rtdcom.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h86tahp2.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 11 +++++------
 1 file changed, 5 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e176b972e4e6..d12054ab8741 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -274,7 +274,6 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd,
 			      struct snd_soc_component *component)
 {
 	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_rtdcom_list *new_rtdcom;
 
 	for_each_rtdcom(rtd, rtdcom) {
 		/* already connected */
@@ -282,14 +281,14 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd,
 			return 0;
 	}
 
-	new_rtdcom = kmalloc(sizeof(*new_rtdcom), GFP_KERNEL);
-	if (!new_rtdcom)
+	rtdcom = kmalloc(sizeof(*rtdcom), GFP_KERNEL);
+	if (!rtdcom)
 		return -ENOMEM;
 
-	new_rtdcom->component = component;
-	INIT_LIST_HEAD(&new_rtdcom->list);
+	rtdcom->component = component;
+	INIT_LIST_HEAD(&rtdcom->list);
 
-	list_add_tail(&new_rtdcom->list, &rtd->component_list);
+	list_add_tail(&rtdcom->list, &rtd->component_list);
 
 	return 0;
 }
-- 
cgit v1.2.3-59-g8ed1b


From b8ba3b572c70c8559ef76c1e0ca02e7b05126e8c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:30:53 +0900
Subject: ASoC: soc-core: tidyup for snd_soc_dapm_new_controls()

snd_soc_dapm_new_controls() registers controls by using
for(... i < num; ...). It means if widget was NULL, num should be zero.
Thus, we don't need to check about widget pointer.
This patch also cares missing return value.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ftmdahow.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d12054ab8741..bb1e9e2c4ff4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1264,16 +1264,14 @@ static int soc_probe_component(struct snd_soc_card *card,
 
 	soc_init_component_debugfs(component);
 
-	if (component->driver->dapm_widgets) {
-		ret = snd_soc_dapm_new_controls(dapm,
+	ret = snd_soc_dapm_new_controls(dapm,
 					component->driver->dapm_widgets,
 					component->driver->num_dapm_widgets);
 
-		if (ret != 0) {
-			dev_err(component->dev,
-				"Failed to create new controls %d\n", ret);
-			goto err_probe;
-		}
+	if (ret != 0) {
+		dev_err(component->dev,
+			"Failed to create new controls %d\n", ret);
+		goto err_probe;
 	}
 
 	for_each_component_dais(component, dai) {
@@ -1990,13 +1988,15 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
 #endif
 
-	if (card->dapm_widgets)
-		snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
-					  card->num_dapm_widgets);
+	ret = snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
+					card->num_dapm_widgets);
+	if (ret < 0)
+		goto probe_end;
 
-	if (card->of_dapm_widgets)
-		snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets,
-					  card->num_of_dapm_widgets);
+	ret = snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets,
+					card->num_of_dapm_widgets);
+	if (ret < 0)
+		goto probe_end;
 
 	/* initialise the sound card only once */
 	if (card->probe) {
-- 
cgit v1.2.3-59-g8ed1b


From abf31feea26c0f412a191c83f408311a0de7435c Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 23 Jul 2019 16:43:40 +0200
Subject: ASoC: Intel: Update request-reply IPC model

struct ipc_message contains fields: header, tx_data and tx_size which
represent TX i.e. request while RX is represented by rx_data and rx_size
with reply's header equivalent missing.

Reply header may contain some vital information including, but not
limited to, received payload size. Some IPCs have entire payload found
within RX header instead. Content and value of said header is context
dependent and may vary between firmware versions and target platform.
Current model does not allow such IPCs to function at all.

Rather than appending yet another parameter to an already long list of
such for sst_ipc_tx_message_XXXs, declare message container in form of
struct sst_ipc_message and add them to parent's ipc_message declaration.

Align haswell, baytrail and skylake with updated request-reply model and
modify their reply processing functions to save RX header within message
container. Despite the range of changes, status quo is achieved.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190723144341.21339-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/baytrail/sst-baytrail-ipc.c |  65 ++++++-----
 sound/soc/intel/common/sst-ipc.c            |  69 ++++++------
 sound/soc/intel/common/sst-ipc.h            |  27 ++---
 sound/soc/intel/haswell/sst-haswell-ipc.c   | 164 +++++++++++++++-------------
 sound/soc/intel/skylake/cnl-sst.c           |   6 +-
 sound/soc/intel/skylake/skl-sst-ipc.c       | 110 +++++++++++--------
 6 files changed, 239 insertions(+), 202 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c
index 8bd1eddcc091..74274bd38f7a 100644
--- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c
+++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c
@@ -211,7 +211,7 @@ static struct sst_byt_stream *sst_byt_get_stream(struct sst_byt *byt,
 static void sst_byt_stream_update(struct sst_byt *byt, struct ipc_message *msg)
 {
 	struct sst_byt_stream *stream;
-	u64 header = msg->header;
+	u64 header = msg->tx.header;
 	u8 stream_id = sst_byt_header_str_id(header);
 	u8 stream_msg = sst_byt_header_msg_id(header);
 
@@ -240,9 +240,10 @@ static int sst_byt_process_reply(struct sst_byt *byt, u64 header)
 	if (msg == NULL)
 		return 1;
 
+	msg->rx.header = header;
 	if (header & IPC_HEADER_LARGE(true)) {
-		msg->rx_size = sst_byt_header_data(header);
-		sst_dsp_inbox_read(byt->dsp, msg->rx_data, msg->rx_size);
+		msg->rx.size = sst_byt_header_data(header);
+		sst_dsp_inbox_read(byt->dsp, msg->rx.data, msg->rx.size);
 	}
 
 	/* update any stream states */
@@ -407,17 +408,18 @@ int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream,
 
 int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream)
 {
-	struct sst_byt_alloc_params *str_req = &stream->request;
-	struct sst_byt_alloc_response *reply = &stream->reply;
-	u64 header;
+	struct sst_ipc_message request, reply = {0};
 	int ret;
 
-	header = sst_byt_header(IPC_IA_ALLOC_STREAM,
-				sizeof(*str_req) + sizeof(u32),
+	request.header = sst_byt_header(IPC_IA_ALLOC_STREAM,
+				sizeof(stream->request) + sizeof(u32),
 				true, stream->str_id);
-	ret = sst_ipc_tx_message_wait(&byt->ipc, header, str_req,
-				      sizeof(*str_req),
-				      reply, sizeof(*reply));
+	request.data = &stream->request;
+	request.size = sizeof(stream->request);
+	reply.data = &stream->reply;
+	reply.size = sizeof(stream->reply);
+
+	ret = sst_ipc_tx_message_wait(&byt->ipc, request, &reply);
 	if (ret < 0) {
 		dev_err(byt->dev, "ipc: error stream commit failed\n");
 		return ret;
@@ -430,7 +432,7 @@ int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream)
 
 int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream)
 {
-	u64 header;
+	struct sst_ipc_message request = {0};
 	int ret = 0;
 	struct sst_dsp *sst = byt->dsp;
 	unsigned long flags;
@@ -438,8 +440,9 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream)
 	if (!stream->commited)
 		goto out;
 
-	header = sst_byt_header(IPC_IA_FREE_STREAM, 0, false, stream->str_id);
-	ret = sst_ipc_tx_message_wait(&byt->ipc, header, NULL, 0, NULL, 0);
+	request.header = sst_byt_header(IPC_IA_FREE_STREAM,
+			0, false, stream->str_id);
+	ret = sst_ipc_tx_message_wait(&byt->ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(byt->dev, "ipc: free stream %d failed\n",
 			stream->str_id);
@@ -459,15 +462,13 @@ out:
 static int sst_byt_stream_operations(struct sst_byt *byt, int type,
 				     int stream_id, int wait)
 {
-	u64 header;
+	struct sst_ipc_message request = {0};
 
-	header = sst_byt_header(type, 0, false, stream_id);
+	request.header = sst_byt_header(type, 0, false, stream_id);
 	if (wait)
-		return sst_ipc_tx_message_wait(&byt->ipc, header, NULL,
-						0, NULL, 0);
+		return sst_ipc_tx_message_wait(&byt->ipc, request, NULL);
 	else
-		return sst_ipc_tx_message_nowait(&byt->ipc, header,
-						NULL, 0);
+		return sst_ipc_tx_message_nowait(&byt->ipc, request);
 }
 
 /* stream ALSA trigger operations */
@@ -475,19 +476,17 @@ int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream,
 			 u32 start_offset)
 {
 	struct sst_byt_start_stream_params start_stream;
-	void *tx_msg;
-	size_t size;
-	u64 header;
+	struct sst_ipc_message request;
 	int ret;
 
 	start_stream.byte_offset = start_offset;
-	header = sst_byt_header(IPC_IA_START_STREAM,
+	request.header = sst_byt_header(IPC_IA_START_STREAM,
 				sizeof(start_stream) + sizeof(u32),
 				true, stream->str_id);
-	tx_msg = &start_stream;
-	size = sizeof(start_stream);
+	request.data = &start_stream;
+	request.size = sizeof(start_stream);
 
-	ret = sst_ipc_tx_message_nowait(&byt->ipc, header, tx_msg, size);
+	ret = sst_ipc_tx_message_nowait(&byt->ipc, request);
 	if (ret < 0)
 		dev_err(byt->dev, "ipc: error failed to start stream %d\n",
 			stream->str_id);
@@ -623,10 +622,10 @@ EXPORT_SYMBOL_GPL(sst_byt_dsp_wait_for_ready);
 
 static void byt_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg)
 {
-	if (msg->header & IPC_HEADER_LARGE(true))
-		sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size);
+	if (msg->tx.header & IPC_HEADER_LARGE(true))
+		sst_dsp_outbox_write(ipc->dsp, msg->tx.data, msg->tx.size);
 
-	sst_dsp_shim_write64_unlocked(ipc->dsp, SST_IPCX, msg->header);
+	sst_dsp_shim_write64_unlocked(ipc->dsp, SST_IPCX, msg->tx.header);
 }
 
 static void byt_shim_dbg(struct sst_generic_ipc *ipc, const char *text)
@@ -648,9 +647,9 @@ static void byt_tx_data_copy(struct ipc_message *msg, char *tx_data,
 	size_t tx_size)
 {
 	/* msg content = lower 32-bit of the header + data */
-	*(u32 *)msg->tx_data = (u32)(msg->header & (u32)-1);
-	memcpy(msg->tx_data + sizeof(u32), tx_data, tx_size);
-	msg->tx_size += sizeof(u32);
+	*(u32 *)msg->tx.data = (u32)(msg->tx.header & (u32)-1);
+	memcpy(msg->tx.data + sizeof(u32), tx_data, tx_size);
+	msg->tx.size += sizeof(u32);
 }
 
 static u64 byt_reply_msg_match(u64 header, u64 *mask)
diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index ef5b66af1cd2..1186a03a88d6 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -43,7 +43,7 @@ static struct ipc_message *msg_get_empty(struct sst_generic_ipc *ipc)
 }
 
 static int tx_wait_done(struct sst_generic_ipc *ipc,
-	struct ipc_message *msg, void *rx_data)
+	struct ipc_message *msg, struct sst_ipc_message *reply)
 {
 	unsigned long flags;
 	int ret;
@@ -62,8 +62,11 @@ static int tx_wait_done(struct sst_generic_ipc *ipc,
 	} else {
 
 		/* copy the data returned from DSP */
-		if (rx_data)
-			memcpy(rx_data, msg->rx_data, msg->rx_size);
+		if (reply) {
+			reply->header = msg->rx.header;
+			if (reply->data)
+				memcpy(reply->data, msg->rx.data, msg->rx.size);
+		}
 		ret = msg->errno;
 	}
 
@@ -72,9 +75,9 @@ static int tx_wait_done(struct sst_generic_ipc *ipc,
 	return ret;
 }
 
-static int ipc_tx_message(struct sst_generic_ipc *ipc, u64 header,
-	void *tx_data, size_t tx_bytes, void *rx_data,
-	size_t rx_bytes, int wait)
+static int ipc_tx_message(struct sst_generic_ipc *ipc,
+	struct sst_ipc_message request,
+	struct sst_ipc_message *reply, int wait)
 {
 	struct ipc_message *msg;
 	unsigned long flags;
@@ -87,23 +90,24 @@ static int ipc_tx_message(struct sst_generic_ipc *ipc, u64 header,
 		return -EBUSY;
 	}
 
-	msg->header = header;
-	msg->tx_size = tx_bytes;
-	msg->rx_size = rx_bytes;
+	msg->tx.header = request.header;
+	msg->tx.size = request.size;
+	msg->rx.header = 0;
+	msg->rx.size = reply ? reply->size : 0;
 	msg->wait = wait;
 	msg->errno = 0;
 	msg->pending = false;
 	msg->complete = false;
 
-	if ((tx_bytes) && (ipc->ops.tx_data_copy != NULL))
-		ipc->ops.tx_data_copy(msg, tx_data, tx_bytes);
+	if ((request.size) && (ipc->ops.tx_data_copy != NULL))
+		ipc->ops.tx_data_copy(msg, request.data, request.size);
 
 	list_add_tail(&msg->list, &ipc->tx_list);
 	schedule_work(&ipc->kwork);
 	spin_unlock_irqrestore(&ipc->dsp->spinlock, flags);
 
 	if (wait)
-		return tx_wait_done(ipc, msg, rx_data);
+		return tx_wait_done(ipc, msg, reply);
 	else
 		return 0;
 }
@@ -118,13 +122,13 @@ static int msg_empty_list_init(struct sst_generic_ipc *ipc)
 		return -ENOMEM;
 
 	for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
-		ipc->msg[i].tx_data = kzalloc(ipc->tx_data_max_size, GFP_KERNEL);
-		if (ipc->msg[i].tx_data == NULL)
+		ipc->msg[i].tx.data = kzalloc(ipc->tx_data_max_size, GFP_KERNEL);
+		if (ipc->msg[i].tx.data == NULL)
 			goto free_mem;
 
-		ipc->msg[i].rx_data = kzalloc(ipc->rx_data_max_size, GFP_KERNEL);
-		if (ipc->msg[i].rx_data == NULL) {
-			kfree(ipc->msg[i].tx_data);
+		ipc->msg[i].rx.data = kzalloc(ipc->rx_data_max_size, GFP_KERNEL);
+		if (ipc->msg[i].rx.data == NULL) {
+			kfree(ipc->msg[i].tx.data);
 			goto free_mem;
 		}
 
@@ -136,8 +140,8 @@ static int msg_empty_list_init(struct sst_generic_ipc *ipc)
 
 free_mem:
 	while (i > 0) {
-		kfree(ipc->msg[i-1].tx_data);
-		kfree(ipc->msg[i-1].rx_data);
+		kfree(ipc->msg[i-1].tx.data);
+		kfree(ipc->msg[i-1].rx.data);
 		--i;
 	}
 	kfree(ipc->msg);
@@ -173,8 +177,8 @@ static void ipc_tx_msgs(struct work_struct *work)
 	spin_unlock_irq(&ipc->dsp->spinlock);
 }
 
-int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc, u64 header,
-	void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes)
+int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc,
+	struct sst_ipc_message request, struct sst_ipc_message *reply)
 {
 	int ret;
 
@@ -187,8 +191,7 @@ int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc, u64 header,
 		if (ipc->ops.check_dsp_lp_on(ipc->dsp, true))
 			return -EIO;
 
-	ret = ipc_tx_message(ipc, header, tx_data, tx_bytes,
-		rx_data, rx_bytes, 1);
+	ret = ipc_tx_message(ipc, request, reply, 1);
 
 	if (ipc->ops.check_dsp_lp_on)
 		if (ipc->ops.check_dsp_lp_on(ipc->dsp, false))
@@ -198,19 +201,17 @@ int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc, u64 header,
 }
 EXPORT_SYMBOL_GPL(sst_ipc_tx_message_wait);
 
-int sst_ipc_tx_message_nowait(struct sst_generic_ipc *ipc, u64 header,
-	void *tx_data, size_t tx_bytes)
+int sst_ipc_tx_message_nowait(struct sst_generic_ipc *ipc,
+	struct sst_ipc_message request)
 {
-	return ipc_tx_message(ipc, header, tx_data, tx_bytes,
-		NULL, 0, 0);
+	return ipc_tx_message(ipc, request, NULL, 0);
 }
 EXPORT_SYMBOL_GPL(sst_ipc_tx_message_nowait);
 
-int sst_ipc_tx_message_nopm(struct sst_generic_ipc *ipc, u64 header,
-	void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes)
+int sst_ipc_tx_message_nopm(struct sst_generic_ipc *ipc,
+	struct sst_ipc_message request, struct sst_ipc_message *reply)
 {
-	return ipc_tx_message(ipc, header, tx_data, tx_bytes,
-		rx_data, rx_bytes, 1);
+	return ipc_tx_message(ipc, request, reply, 1);
 }
 EXPORT_SYMBOL_GPL(sst_ipc_tx_message_nopm);
 
@@ -230,7 +231,7 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc,
 	}
 
 	list_for_each_entry(msg, &ipc->rx_list, list) {
-		if ((msg->header & mask) == header)
+		if ((msg->tx.header & mask) == header)
 			return msg;
 	}
 
@@ -304,8 +305,8 @@ void sst_ipc_fini(struct sst_generic_ipc *ipc)
 
 	if (ipc->msg) {
 		for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
-			kfree(ipc->msg[i].tx_data);
-			kfree(ipc->msg[i].rx_data);
+			kfree(ipc->msg[i].tx.data);
+			kfree(ipc->msg[i].rx.data);
 		}
 		kfree(ipc->msg);
 	}
diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h
index ef38600e88f7..08c4831b2664 100644
--- a/sound/soc/intel/common/sst-ipc.h
+++ b/sound/soc/intel/common/sst-ipc.h
@@ -17,15 +17,16 @@
 
 #define IPC_MAX_MAILBOX_BYTES	256
 
-struct ipc_message {
-	struct list_head list;
+struct sst_ipc_message {
 	u64 header;
+	void *data;
+	size_t size;
+};
 
-	/* direction wrt host CPU */
-	char *tx_data;
-	size_t tx_size;
-	char *rx_data;
-	size_t rx_size;
+struct ipc_message {
+	struct list_head list;
+	struct sst_ipc_message tx;
+	struct sst_ipc_message rx;
 
 	wait_queue_head_t waitq;
 	bool pending;
@@ -66,14 +67,14 @@ struct sst_generic_ipc {
 	struct sst_plat_ipc_ops ops;
 };
 
-int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc, u64 header,
-	void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes);
+int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc,
+	struct sst_ipc_message request, struct sst_ipc_message *reply);
 
-int sst_ipc_tx_message_nowait(struct sst_generic_ipc *ipc, u64 header,
-	void *tx_data, size_t tx_bytes);
+int sst_ipc_tx_message_nowait(struct sst_generic_ipc *ipc,
+	struct sst_ipc_message request);
 
-int sst_ipc_tx_message_nopm(struct sst_generic_ipc *ipc, u64 header,
-	void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes);
+int sst_ipc_tx_message_nopm(struct sst_generic_ipc *ipc,
+	struct sst_ipc_message request, struct sst_ipc_message *reply);
 
 struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc,
 	u64 header);
diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index a83b92d6bea8..5c73b11375e3 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -511,7 +511,7 @@ static void hsw_notification_work(struct work_struct *work)
 static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg)
 {
 	struct sst_hsw_stream *stream;
-	u32 header = msg->header & ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK);
+	u32 header = msg->tx.header & ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK);
 	u32 stream_id = msg_get_stream_id(header);
 	u32 stream_msg = msg_get_stream_type(header);
 
@@ -552,6 +552,7 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
 		return -EIO;
 	}
 
+	msg->rx.header = header;
 	/* first process the header */
 	switch (reply) {
 	case IPC_GLB_REPLY_PENDING:
@@ -562,13 +563,13 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
 	case IPC_GLB_REPLY_SUCCESS:
 		if (msg->pending) {
 			trace_ipc_pending_reply("completed", header);
-			sst_dsp_inbox_read(hsw->dsp, msg->rx_data,
-				msg->rx_size);
+			sst_dsp_inbox_read(hsw->dsp, msg->rx.data,
+				msg->rx.size);
 			hsw->ipc.pending = false;
 		} else {
 			/* copy data from the DSP */
-			sst_dsp_outbox_read(hsw->dsp, msg->rx_data,
-				msg->rx_size);
+			sst_dsp_outbox_read(hsw->dsp, msg->rx.data,
+				msg->rx.size);
 		}
 		break;
 	/* these will be rare - but useful for debug */
@@ -810,11 +811,13 @@ static irqreturn_t hsw_irq_thread(int irq, void *context)
 int sst_hsw_fw_get_version(struct sst_hsw *hsw,
 	struct sst_hsw_ipc_fw_version *version)
 {
+	struct sst_ipc_message request = {0}, reply = {0};
 	int ret;
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc,
-		IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION),
-		NULL, 0, version, sizeof(*version));
+	request.header = IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION);
+	reply.data = version;
+	reply.size = sizeof(*version);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, &reply);
 	if (ret < 0)
 		dev_err(hsw->dev, "error: get version failed\n");
 
@@ -840,7 +843,7 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume)
 {
 	struct sst_hsw_ipc_volume_req *req;
-	u32 header;
+	struct sst_ipc_message request;
 	int ret;
 
 	trace_ipc_request("set stream volume", stream->reply.stream_hw_id);
@@ -848,11 +851,11 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 	if (channel >= 2 && channel != SST_HSW_CHANNELS_ALL)
 		return -EINVAL;
 
-	header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+	request.header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
 		IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
-	header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
-	header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
-	header |= (stage_id << IPC_STG_ID_SHIFT);
+	request.header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
+	request.header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
+	request.header |= (stage_id << IPC_STG_ID_SHIFT);
 
 	req = &stream->vol_req;
 	req->target_volume = volume;
@@ -877,8 +880,9 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
 		req->channel = channel;
 	}
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header, req,
-		sizeof(*req), NULL, 0);
+	request.data = req;
+	request.size = sizeof(*req);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(hsw->dev, "error: set stream volume failed\n");
 		return ret;
@@ -905,7 +909,7 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 	u32 volume)
 {
 	struct sst_hsw_ipc_volume_req req;
-	u32 header;
+	struct sst_ipc_message request;
 	int ret;
 
 	trace_ipc_request("set mixer volume", volume);
@@ -933,18 +937,19 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
 		req.channel = channel;
 	}
 
-	header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+	request.header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
 		IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
-	header |= (hsw->mixer_info.mixer_hw_id << IPC_STR_ID_SHIFT);
-	header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
-	header |= (stage_id << IPC_STG_ID_SHIFT);
+	request.header |= (hsw->mixer_info.mixer_hw_id << IPC_STR_ID_SHIFT);
+	request.header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
+	request.header |= (stage_id << IPC_STG_ID_SHIFT);
 
 	req.curve_duration = hsw->curve_duration;
 	req.curve_type = hsw->curve_type;
 	req.target_volume = volume;
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &req,
-		sizeof(req), NULL, 0);
+	request.data = &req;
+	request.size = sizeof(req);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(hsw->dev, "error: set mixer volume failed\n");
 		return ret;
@@ -983,7 +988,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
 
 int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 {
-	u32 header;
+	struct sst_ipc_message request;
 	int ret = 0;
 	struct sst_dsp *sst = hsw->dsp;
 	unsigned long flags;
@@ -1000,10 +1005,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 	trace_ipc_request("stream free", stream->host_id);
 
 	stream->free_req.stream_id = stream->reply.stream_hw_id;
-	header = IPC_GLB_TYPE(IPC_GLB_FREE_STREAM);
+	request.header = IPC_GLB_TYPE(IPC_GLB_FREE_STREAM);
+	request.data = &stream->free_req;
+	request.size = sizeof(stream->free_req);
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &stream->free_req,
-		sizeof(stream->free_req), NULL, 0);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(hsw->dev, "error: free stream %d failed\n",
 			stream->free_req.stream_id);
@@ -1175,9 +1181,7 @@ int sst_hsw_stream_set_module_info(struct sst_hsw *hsw,
 
 int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 {
-	struct sst_hsw_ipc_stream_alloc_req *str_req = &stream->request;
-	struct sst_hsw_ipc_stream_alloc_reply *reply = &stream->reply;
-	u32 header;
+	struct sst_ipc_message request, reply = {0};
 	int ret;
 
 	if (!stream) {
@@ -1192,10 +1196,13 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 
 	trace_ipc_request("stream alloc", stream->host_id);
 
-	header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
+	request.header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
+	request.data = &stream->request;
+	request.size = sizeof(stream->request);
+	reply.data = &stream->reply;
+	reply.size = sizeof(stream->reply);
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header, str_req,
-		sizeof(*str_req), reply, sizeof(*reply));
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, &reply);
 	if (ret < 0) {
 		dev_err(hsw->dev, "error: stream commit failed\n");
 		return ret;
@@ -1235,23 +1242,22 @@ void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw,
  ABI to be opaque to client PCM drivers to cope with any future ABI changes */
 int sst_hsw_mixer_get_info(struct sst_hsw *hsw)
 {
-	struct sst_hsw_ipc_stream_info_reply *reply;
-	u32 header;
+	struct sst_ipc_message request = {0}, reply = {0};
 	int ret;
 
-	reply = &hsw->mixer_info;
-	header = IPC_GLB_TYPE(IPC_GLB_GET_MIXER_STREAM_INFO);
+	request.header = IPC_GLB_TYPE(IPC_GLB_GET_MIXER_STREAM_INFO);
+	reply.data = &hsw->mixer_info;
+	reply.size = sizeof(hsw->mixer_info);
 
 	trace_ipc_request("get global mixer info", 0);
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header, NULL, 0,
-		reply, sizeof(*reply));
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, &reply);
 	if (ret < 0) {
 		dev_err(hsw->dev, "error: get stream info failed\n");
 		return ret;
 	}
 
-	trace_hsw_mixer_info_reply(reply);
+	trace_hsw_mixer_info_reply(&hsw->mixer_info);
 
 	return 0;
 }
@@ -1260,16 +1266,15 @@ int sst_hsw_mixer_get_info(struct sst_hsw *hsw)
 static int sst_hsw_stream_operations(struct sst_hsw *hsw, int type,
 	int stream_id, int wait)
 {
-	u32 header;
+	struct sst_ipc_message request = {0};
 
-	header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(type);
-	header |= (stream_id << IPC_STR_ID_SHIFT);
+	request.header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE);
+	request.header |= IPC_STR_TYPE(type) | (stream_id << IPC_STR_ID_SHIFT);
 
 	if (wait)
-		return sst_ipc_tx_message_wait(&hsw->ipc, header,
-			NULL, 0, NULL, 0);
+		return sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	else
-		return sst_ipc_tx_message_nowait(&hsw->ipc, header, NULL, 0);
+		return sst_ipc_tx_message_nowait(&hsw->ipc, request);
 }
 
 /* Stream ALSA trigger operations */
@@ -1377,8 +1382,8 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
 	enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
 	enum sst_hsw_device_mode mode, u32 clock_divider)
 {
+	struct sst_ipc_message request;
 	struct sst_hsw_ipc_device_config_req config;
-	u32 header;
 	int ret;
 
 	trace_ipc_request("set device config", dev);
@@ -1394,10 +1399,11 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
 
 	trace_hsw_device_config_req(&config);
 
-	header = IPC_GLB_TYPE(IPC_GLB_SET_DEVICE_FORMATS);
+	request.header = IPC_GLB_TYPE(IPC_GLB_SET_DEVICE_FORMATS);
+	request.data = &config;
+	request.size = sizeof(config);
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &config,
-		sizeof(config), NULL, 0);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	if (ret < 0)
 		dev_err(hsw->dev, "error: set device formats failed\n");
 
@@ -1409,16 +1415,20 @@ EXPORT_SYMBOL_GPL(sst_hsw_device_set_config);
 int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 	enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
 {
-	u32 header, state_;
+	struct sst_ipc_message request, reply = {0};
+	u32 state_;
 	int ret, item;
 
-	header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
 	state_ = state;
+	request.header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
+	request.data = &state_;
+	request.size = sizeof(state_);
+	reply.data = dx;
+	reply.size = sizeof(*dx);
 
 	trace_ipc_request("PM enter Dx state", state);
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &state_,
-		sizeof(state_), dx, sizeof(*dx));
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, &reply);
 	if (ret < 0) {
 		dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state);
 		return ret;
@@ -1878,7 +1888,7 @@ int sst_hsw_module_enable(struct sst_hsw *hsw,
 	u32 module_id, u32 instance_id)
 {
 	int ret;
-	u32 header = 0;
+	struct sst_ipc_message request;
 	struct sst_hsw_ipc_module_config config;
 	struct sst_module *module;
 	struct sst_module_runtime *runtime;
@@ -1907,10 +1917,10 @@ int sst_hsw_module_enable(struct sst_hsw *hsw,
 		return -ENXIO;
 	}
 
-	header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) |
+	request.header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) |
 			IPC_MODULE_OPERATION(IPC_MODULE_ENABLE) |
 			IPC_MODULE_ID(module_id);
-	dev_dbg(dev, "module enable header: %x\n", header);
+	dev_dbg(dev, "module enable header: %x\n", (u32)request.header);
 
 	config.map.module_entries_count = 1;
 	config.map.module_entries[0].module_id = module->id;
@@ -1932,8 +1942,9 @@ int sst_hsw_module_enable(struct sst_hsw *hsw,
 		config.scratch_mem.size, config.scratch_mem.offset,
 		config.map.module_entries[0].entry_point);
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header,
-			&config, sizeof(config), NULL, 0);
+	request.data = &config;
+	request.size = sizeof(config);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	if (ret < 0)
 		dev_err(dev, "ipc: module enable failed - %d\n", ret);
 	else
@@ -1946,7 +1957,7 @@ int sst_hsw_module_disable(struct sst_hsw *hsw,
 	u32 module_id, u32 instance_id)
 {
 	int ret;
-	u32 header;
+	struct sst_ipc_message request = {0};
 	struct sst_module *module;
 	struct device *dev = hsw->dev;
 	struct sst_dsp *dsp = hsw->dsp;
@@ -1967,11 +1978,11 @@ int sst_hsw_module_disable(struct sst_hsw *hsw,
 		return -ENXIO;
 	}
 
-	header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) |
+	request.header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) |
 			IPC_MODULE_OPERATION(IPC_MODULE_DISABLE) |
 			IPC_MODULE_ID(module_id);
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header,  NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	if (ret < 0)
 		dev_err(dev, "module disable failed - %d\n", ret);
 	else
@@ -1985,15 +1996,16 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw,
 	u32 param_size, char *param)
 {
 	int ret;
-	u32 header = 0;
-	u32 payload_size = 0, transfer_parameter_size = 0;
+	struct sst_ipc_message request = {0};
+	u32 payload_size = 0;
 	struct sst_hsw_transfer_parameter *parameter;
 	struct device *dev = hsw->dev;
 
-	header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) |
+	request.header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) |
 			IPC_MODULE_OPERATION(IPC_MODULE_SET_PARAMETER) |
 			IPC_MODULE_ID(module_id);
-	dev_dbg(dev, "sst_hsw_module_set_param header=%x\n", header);
+	dev_dbg(dev, "sst_hsw_module_set_param header=%x\n",
+			(u32)request.header);
 
 	payload_size = param_size +
 		sizeof(struct sst_hsw_transfer_parameter) -
@@ -2003,14 +2015,14 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw,
 
 	if (payload_size <= SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE) {
 		/* short parameter, mailbox can contain data */
-		dev_dbg(dev, "transfer parameter size : %d\n",
-			transfer_parameter_size);
+		dev_dbg(dev, "transfer parameter size : %lu\n",
+			request.size);
 
-		transfer_parameter_size = ALIGN(payload_size, 4);
-		dev_dbg(dev, "transfer parameter aligned size : %d\n",
-			transfer_parameter_size);
+		request.size = ALIGN(payload_size, 4);
+		dev_dbg(dev, "transfer parameter aligned size : %lu\n",
+			request.size);
 
-		parameter = kzalloc(transfer_parameter_size, GFP_KERNEL);
+		parameter = kzalloc(request.size, GFP_KERNEL);
 		if (parameter == NULL)
 			return -ENOMEM;
 
@@ -2022,9 +2034,9 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw,
 
 	parameter->parameter_id = parameter_id;
 	parameter->data_size = param_size;
+	request.data = parameter;
 
-	ret = sst_ipc_tx_message_wait(&hsw->ipc, header,
-		parameter, transfer_parameter_size , NULL, 0);
+	ret = sst_ipc_tx_message_wait(&hsw->ipc, request, NULL);
 	if (ret < 0)
 		dev_err(dev, "ipc: module set parameter failed - %d\n", ret);
 
@@ -2041,8 +2053,8 @@ static struct sst_dsp_device hsw_dev = {
 static void hsw_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg)
 {
 	/* send the message */
-	sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size);
-	sst_dsp_ipc_msg_tx(ipc->dsp, msg->header);
+	sst_dsp_outbox_write(ipc->dsp, msg->tx.data, msg->tx.size);
+	sst_dsp_ipc_msg_tx(ipc->dsp, msg->tx.header);
 }
 
 static void hsw_shim_dbg(struct sst_generic_ipc *ipc, const char *text)
@@ -2063,7 +2075,7 @@ static void hsw_shim_dbg(struct sst_generic_ipc *ipc, const char *text)
 static void hsw_tx_data_copy(struct ipc_message *msg, char *tx_data,
 	size_t tx_size)
 {
-	memcpy(msg->tx_data, tx_data, tx_size);
+	memcpy(msg->tx.data, tx_data, tx_size);
 }
 
 static u64 hsw_reply_msg_match(u64 header, u64 *mask)
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
index 2d748a335bcf..4f64f097e9ae 100644
--- a/sound/soc/intel/skylake/cnl-sst.c
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -366,10 +366,10 @@ static struct sst_dsp_device cnl_dev = {
 
 static void cnl_ipc_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg)
 {
-	struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->header);
+	struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->tx.header);
 
-	if (msg->tx_size)
-		sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size);
+	if (msg->tx.size)
+		sst_dsp_outbox_write(ipc->dsp, msg->tx.data, msg->tx.size);
 	sst_dsp_shim_write_unlocked(ipc->dsp, CNL_ADSP_REG_HIPCIDD,
 				    header->extension);
 	sst_dsp_shim_write_unlocked(ipc->dsp, CNL_ADSP_REG_HIPCIDR,
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index ee1493acc9a8..a2b69a02aab2 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -281,7 +281,7 @@ void skl_ipc_tx_data_copy(struct ipc_message *msg, char *tx_data,
 		size_t tx_size)
 {
 	if (tx_size)
-		memcpy(msg->tx_data, tx_data, tx_size);
+		memcpy(msg->tx.data, tx_data, tx_size);
 }
 
 static bool skl_ipc_is_dsp_busy(struct sst_dsp *dsp)
@@ -295,10 +295,10 @@ static bool skl_ipc_is_dsp_busy(struct sst_dsp *dsp)
 /* Lock to be held by caller */
 static void skl_ipc_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg)
 {
-	struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->header);
+	struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->tx.header);
 
-	if (msg->tx_size)
-		sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size);
+	if (msg->tx.size)
+		sst_dsp_outbox_write(ipc->dsp, msg->tx.data, msg->tx.size);
 	sst_dsp_shim_write_unlocked(ipc->dsp, SKL_ADSP_REG_HIPCIE,
 						header->extension);
 	sst_dsp_shim_write_unlocked(ipc->dsp, SKL_ADSP_REG_HIPCI,
@@ -447,11 +447,12 @@ void skl_ipc_process_reply(struct sst_generic_ipc *ipc,
 		return;
 	}
 
+	msg->rx.header = *ipc_header;
 	/* first process the header */
 	if (reply == IPC_GLB_REPLY_SUCCESS) {
 		dev_dbg(ipc->dev, "ipc FW reply %x: success\n", header.primary);
 		/* copy the rx data from the mailbox */
-		sst_dsp_inbox_read(ipc->dsp, msg->rx_data, msg->rx_size);
+		sst_dsp_inbox_read(ipc->dsp, msg->rx.data, msg->rx.size);
 		switch (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) {
 		case IPC_GLB_LOAD_MULTIPLE_MODS:
 		case IPC_GLB_LOAD_LIBRARY:
@@ -635,7 +636,7 @@ int skl_ipc_create_pipeline(struct sst_generic_ipc *ipc,
 		u16 ppl_mem_size, u8 ppl_type, u8 instance_id, u8 lp_mode)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
@@ -646,9 +647,10 @@ int skl_ipc_create_pipeline(struct sst_generic_ipc *ipc,
 	header.primary |= IPC_PPL_MEM_SIZE(ppl_mem_size);
 
 	header.extension = IPC_PPL_LP_MODE(lp_mode);
+	request.header = *(u64 *)(&header);
 
 	dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: create pipeline fail, err: %d\n", ret);
 		return ret;
@@ -661,16 +663,17 @@ EXPORT_SYMBOL_GPL(skl_ipc_create_pipeline);
 int skl_ipc_delete_pipeline(struct sst_generic_ipc *ipc, u8 instance_id)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
 	header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST);
 	header.primary |= IPC_GLB_TYPE(IPC_GLB_DELETE_PPL);
 	header.primary |= IPC_INSTANCE_ID(instance_id);
+	request.header = *(u64 *)(&header);
 
 	dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: delete pipeline failed, err %d\n", ret);
 		return ret;
@@ -684,7 +687,7 @@ int skl_ipc_set_pipeline_state(struct sst_generic_ipc *ipc,
 		u8 instance_id, enum skl_ipc_pipeline_state state)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
@@ -692,9 +695,10 @@ int skl_ipc_set_pipeline_state(struct sst_generic_ipc *ipc,
 	header.primary |= IPC_GLB_TYPE(IPC_GLB_SET_PPL_STATE);
 	header.primary |= IPC_INSTANCE_ID(instance_id);
 	header.primary |= IPC_PPL_STATE(state);
+	request.header = *(u64 *)(&header);
 
 	dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: set pipeline state failed, err: %d\n", ret);
 		return ret;
@@ -707,7 +711,7 @@ int
 skl_ipc_save_pipeline(struct sst_generic_ipc *ipc, u8 instance_id, int dma_id)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
@@ -716,8 +720,10 @@ skl_ipc_save_pipeline(struct sst_generic_ipc *ipc, u8 instance_id, int dma_id)
 	header.primary |= IPC_INSTANCE_ID(instance_id);
 
 	header.extension = IPC_DMA_ID(dma_id);
+	request.header = *(u64 *)(&header);
+
 	dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: save pipeline failed, err: %d\n", ret);
 		return ret;
@@ -730,16 +736,17 @@ EXPORT_SYMBOL_GPL(skl_ipc_save_pipeline);
 int skl_ipc_restore_pipeline(struct sst_generic_ipc *ipc, u8 instance_id)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
 	header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST);
 	header.primary |= IPC_GLB_TYPE(IPC_GLB_RESTORE_PPL);
 	header.primary |= IPC_INSTANCE_ID(instance_id);
+	request.header = *(u64 *)(&header);
 
 	dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: restore  pipeline failed, err: %d\n", ret);
 		return ret;
@@ -753,7 +760,7 @@ int skl_ipc_set_dx(struct sst_generic_ipc *ipc, u8 instance_id,
 		u16 module_id, struct skl_ipc_dxstate_info *dx)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request;
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_MOD_MSG);
@@ -762,10 +769,13 @@ int skl_ipc_set_dx(struct sst_generic_ipc *ipc, u8 instance_id,
 	header.primary |= IPC_MOD_INSTANCE_ID(instance_id);
 	header.primary |= IPC_MOD_ID(module_id);
 
+	request.header = *(u64 *)(&header);
+	request.data = dx;
+	request.size = sizeof(*dx);
+
 	dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__,
 			 header.primary, header.extension);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header,
-				dx, sizeof(*dx), NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: set dx failed, err %d\n", ret);
 		return ret;
@@ -779,7 +789,7 @@ int skl_ipc_init_instance(struct sst_generic_ipc *ipc,
 		struct skl_ipc_init_instance_msg *msg, void *param_data)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request;
 	int ret;
 	u32 *buffer = (u32 *)param_data;
 	 /* param_block_size must be in dwords */
@@ -799,10 +809,13 @@ int skl_ipc_init_instance(struct sst_generic_ipc *ipc,
 	header.extension |= IPC_PARAM_BLOCK_SIZE(param_block_size);
 	header.extension |= IPC_DOMAIN(msg->domain);
 
+	request.header = *(u64 *)(&header);
+	request.data = param_data;
+	request.size = msg->param_data_size;
+
 	dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__,
 			 header.primary, header.extension);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, param_data,
-			msg->param_data_size, NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: init instance failed\n");
@@ -817,7 +830,7 @@ int skl_ipc_bind_unbind(struct sst_generic_ipc *ipc,
 		struct skl_ipc_bind_unbind_msg *msg)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	u8 bind_unbind = msg->bind ? IPC_MOD_BIND : IPC_MOD_UNBIND;
 	int ret;
 
@@ -831,10 +844,11 @@ int skl_ipc_bind_unbind(struct sst_generic_ipc *ipc,
 	header.extension |= IPC_DST_MOD_INSTANCE_ID(msg->dst_instance_id);
 	header.extension |= IPC_DST_QUEUE(msg->dst_queue);
 	header.extension |= IPC_SRC_QUEUE(msg->src_queue);
+	request.header = *(u64 *)(&header);
 
 	dev_dbg(ipc->dev, "In %s hdr=%x ext=%x\n", __func__, header.primary,
 			 header.extension);
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0) {
 		dev_err(ipc->dev, "ipc: bind/unbind failed\n");
 		return ret;
@@ -854,7 +868,7 @@ int skl_ipc_load_modules(struct sst_generic_ipc *ipc,
 				u8 module_cnt, void *data)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request;
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
@@ -862,8 +876,11 @@ int skl_ipc_load_modules(struct sst_generic_ipc *ipc,
 	header.primary |= IPC_GLB_TYPE(IPC_GLB_LOAD_MULTIPLE_MODS);
 	header.primary |= IPC_LOAD_MODULE_CNT(module_cnt);
 
-	ret = sst_ipc_tx_message_nowait(ipc, *ipc_header, data,
-				(sizeof(u16) * module_cnt));
+	request.header = *(u64 *)(&header);
+	request.data = data;
+	request.size = sizeof(u16) * module_cnt;
+
+	ret = sst_ipc_tx_message_nowait(ipc, request);
 	if (ret < 0)
 		dev_err(ipc->dev, "ipc: load modules failed :%d\n", ret);
 
@@ -875,7 +892,7 @@ int skl_ipc_unload_modules(struct sst_generic_ipc *ipc, u8 module_cnt,
 							void *data)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request;
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
@@ -883,8 +900,11 @@ int skl_ipc_unload_modules(struct sst_generic_ipc *ipc, u8 module_cnt,
 	header.primary |= IPC_GLB_TYPE(IPC_GLB_UNLOAD_MULTIPLE_MODS);
 	header.primary |= IPC_LOAD_MODULE_CNT(module_cnt);
 
-	ret = sst_ipc_tx_message_wait(ipc, *ipc_header, data,
-				(sizeof(u16) * module_cnt), NULL, 0);
+	request.header = *(u64 *)(&header);
+	request.data = data;
+	request.size = sizeof(u16) * module_cnt;
+
+	ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	if (ret < 0)
 		dev_err(ipc->dev, "ipc: unload modules failed :%d\n", ret);
 
@@ -896,7 +916,7 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc,
 		struct skl_ipc_large_config_msg *msg, u32 *param)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request;
 	int ret = 0;
 	size_t sz_remaining, tx_size, data_offset;
 
@@ -923,9 +943,11 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc,
 			header.primary, header.extension);
 		dev_dbg(ipc->dev, "transmitting offset: %#x, size: %#x\n",
 			(unsigned)data_offset, (unsigned)tx_size);
-		ret = sst_ipc_tx_message_wait(ipc, *ipc_header,
-					  ((char *)param) + data_offset,
-					  tx_size, NULL, 0);
+
+		request.header = *(u64 *)(&header);
+		request.data = ((char *)param) + data_offset;
+		request.size = tx_size;
+		ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 		if (ret < 0) {
 			dev_err(ipc->dev,
 				"ipc: set large config fail, err: %d\n", ret);
@@ -950,7 +972,7 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
 		struct skl_ipc_large_config_msg *msg, u32 *param)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0}, reply = {0};
 	int ret = 0;
 	size_t sz_remaining, rx_size, data_offset;
 
@@ -974,9 +996,10 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
 		if (rx_size == sz_remaining)
 			header.extension |= IPC_FINAL_BLOCK(1);
 
-		ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0,
-					      ((char *)param) + data_offset,
-					      msg->param_data_size);
+		request.header = *(u64 *)(&header);
+		reply.data = ((char *)param) + data_offset;
+		reply.size = msg->param_data_size;
+		ret = sst_ipc_tx_message_wait(ipc, request, &reply);
 		if (ret < 0) {
 			dev_err(ipc->dev,
 				"ipc: get large config fail, err: %d\n", ret);
@@ -1001,7 +1024,7 @@ int skl_sst_ipc_load_library(struct sst_generic_ipc *ipc,
 				u8 dma_id, u8 table_id, bool wait)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	int ret = 0;
 
 	header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG);
@@ -1009,12 +1032,12 @@ int skl_sst_ipc_load_library(struct sst_generic_ipc *ipc,
 	header.primary |= IPC_GLB_TYPE(IPC_GLB_LOAD_LIBRARY);
 	header.primary |= IPC_MOD_INSTANCE_ID(table_id);
 	header.primary |= IPC_MOD_ID(dma_id);
+	request.header = *(u64 *)(&header);
 
 	if (wait)
-		ret = sst_ipc_tx_message_wait(ipc, *ipc_header,
-					NULL, 0, NULL, 0);
+		ret = sst_ipc_tx_message_wait(ipc, request, NULL);
 	else
-		ret = sst_ipc_tx_message_nowait(ipc, *ipc_header, NULL, 0);
+		ret = sst_ipc_tx_message_nowait(ipc, request);
 
 	if (ret < 0)
 		dev_err(ipc->dev, "ipc: load lib failed\n");
@@ -1026,7 +1049,7 @@ EXPORT_SYMBOL_GPL(skl_sst_ipc_load_library);
 int skl_ipc_set_d0ix(struct sst_generic_ipc *ipc, struct skl_ipc_d0ix_msg *msg)
 {
 	struct skl_ipc_header header = {0};
-	u64 *ipc_header = (u64 *)(&header);
+	struct sst_ipc_message request = {0};
 	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_MOD_MSG);
@@ -1037,6 +1060,7 @@ int skl_ipc_set_d0ix(struct sst_generic_ipc *ipc, struct skl_ipc_d0ix_msg *msg)
 
 	header.extension = IPC_D0IX_WAKE(msg->wake);
 	header.extension |= IPC_D0IX_STREAMING(msg->streaming);
+	request.header = *(u64 *)(&header);
 
 	dev_dbg(ipc->dev, "In %s primary=%x ext=%x\n", __func__,
 			header.primary,	header.extension);
@@ -1044,7 +1068,7 @@ int skl_ipc_set_d0ix(struct sst_generic_ipc *ipc, struct skl_ipc_d0ix_msg *msg)
 	/*
 	 * Use the nopm IPC here as we dont want it checking for D0iX
 	 */
-	ret = sst_ipc_tx_message_nopm(ipc, *ipc_header, NULL, 0, NULL, 0);
+	ret = sst_ipc_tx_message_nopm(ipc, request, NULL);
 	if (ret < 0)
 		dev_err(ipc->dev, "ipc: set d0ix failed, err %d\n", ret);
 
-- 
cgit v1.2.3-59-g8ed1b


From 5f0ac20ed6db1d6da2eea8b862cf3d54fdfb5830 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Tue, 6 Aug 2019 18:12:10 +0300
Subject: ASoC: fsl_sai: Add registers definition for multiple datalines

SAI IP supports up to 8 data lines. The configuration of
supported number of data lines is decided at SoC integration
time.

This patch adds definitions for all related data TX/RX registers:
	* TDR0..7, Transmit data register
	* TFR0..7, Transmit FIFO register
	* RDR0..7, Receive data register
	* RFR0..7, Receive FIFO register

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20190806151214.6783-2-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 76 ++++++++++++++++++++++++++++++++++++++++++-------
 sound/soc/fsl/fsl_sai.h | 36 ++++++++++++++++++++---
 2 files changed, 98 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 8f4d9fa95599..e4221f2a5ee3 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -685,7 +685,14 @@ static struct reg_default fsl_sai_reg_defaults[] = {
 	{FSL_SAI_TCR3, 0},
 	{FSL_SAI_TCR4, 0},
 	{FSL_SAI_TCR5, 0},
-	{FSL_SAI_TDR,  0},
+	{FSL_SAI_TDR0, 0},
+	{FSL_SAI_TDR1, 0},
+	{FSL_SAI_TDR2, 0},
+	{FSL_SAI_TDR3, 0},
+	{FSL_SAI_TDR4, 0},
+	{FSL_SAI_TDR5, 0},
+	{FSL_SAI_TDR6, 0},
+	{FSL_SAI_TDR7, 0},
 	{FSL_SAI_TMR,  0},
 	{FSL_SAI_RCR1, 0},
 	{FSL_SAI_RCR2, 0},
@@ -704,7 +711,14 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
 	case FSL_SAI_TCR3:
 	case FSL_SAI_TCR4:
 	case FSL_SAI_TCR5:
-	case FSL_SAI_TFR:
+	case FSL_SAI_TFR0:
+	case FSL_SAI_TFR1:
+	case FSL_SAI_TFR2:
+	case FSL_SAI_TFR3:
+	case FSL_SAI_TFR4:
+	case FSL_SAI_TFR5:
+	case FSL_SAI_TFR6:
+	case FSL_SAI_TFR7:
 	case FSL_SAI_TMR:
 	case FSL_SAI_RCSR:
 	case FSL_SAI_RCR1:
@@ -712,8 +726,22 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
 	case FSL_SAI_RCR3:
 	case FSL_SAI_RCR4:
 	case FSL_SAI_RCR5:
-	case FSL_SAI_RDR:
-	case FSL_SAI_RFR:
+	case FSL_SAI_RDR0:
+	case FSL_SAI_RDR1:
+	case FSL_SAI_RDR2:
+	case FSL_SAI_RDR3:
+	case FSL_SAI_RDR4:
+	case FSL_SAI_RDR5:
+	case FSL_SAI_RDR6:
+	case FSL_SAI_RDR7:
+	case FSL_SAI_RFR0:
+	case FSL_SAI_RFR1:
+	case FSL_SAI_RFR2:
+	case FSL_SAI_RFR3:
+	case FSL_SAI_RFR4:
+	case FSL_SAI_RFR5:
+	case FSL_SAI_RFR6:
+	case FSL_SAI_RFR7:
 	case FSL_SAI_RMR:
 		return true;
 	default:
@@ -726,9 +754,30 @@ static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
 	switch (reg) {
 	case FSL_SAI_TCSR:
 	case FSL_SAI_RCSR:
-	case FSL_SAI_TFR:
-	case FSL_SAI_RFR:
-	case FSL_SAI_RDR:
+	case FSL_SAI_TFR0:
+	case FSL_SAI_TFR1:
+	case FSL_SAI_TFR2:
+	case FSL_SAI_TFR3:
+	case FSL_SAI_TFR4:
+	case FSL_SAI_TFR5:
+	case FSL_SAI_TFR6:
+	case FSL_SAI_TFR7:
+	case FSL_SAI_RFR0:
+	case FSL_SAI_RFR1:
+	case FSL_SAI_RFR2:
+	case FSL_SAI_RFR3:
+	case FSL_SAI_RFR4:
+	case FSL_SAI_RFR5:
+	case FSL_SAI_RFR6:
+	case FSL_SAI_RFR7:
+	case FSL_SAI_RDR0:
+	case FSL_SAI_RDR1:
+	case FSL_SAI_RDR2:
+	case FSL_SAI_RDR3:
+	case FSL_SAI_RDR4:
+	case FSL_SAI_RDR5:
+	case FSL_SAI_RDR6:
+	case FSL_SAI_RDR7:
 		return true;
 	default:
 		return false;
@@ -744,7 +793,14 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 	case FSL_SAI_TCR3:
 	case FSL_SAI_TCR4:
 	case FSL_SAI_TCR5:
-	case FSL_SAI_TDR:
+	case FSL_SAI_TDR0:
+	case FSL_SAI_TDR1:
+	case FSL_SAI_TDR2:
+	case FSL_SAI_TDR3:
+	case FSL_SAI_TDR4:
+	case FSL_SAI_TDR5:
+	case FSL_SAI_TDR6:
+	case FSL_SAI_TDR7:
 	case FSL_SAI_TMR:
 	case FSL_SAI_RCSR:
 	case FSL_SAI_RCR1:
@@ -883,8 +939,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
 				   MCLK_DIR(index));
 	}
 
-	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
-	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
+	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR0;
+	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR0;
 	sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
 	sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
 
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 7c1ef671da28..4bb478041d67 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -20,8 +20,22 @@
 #define FSL_SAI_TCR3	0x0c /* SAI Transmit Configuration 3 */
 #define FSL_SAI_TCR4	0x10 /* SAI Transmit Configuration 4 */
 #define FSL_SAI_TCR5	0x14 /* SAI Transmit Configuration 5 */
-#define FSL_SAI_TDR	0x20 /* SAI Transmit Data */
-#define FSL_SAI_TFR	0x40 /* SAI Transmit FIFO */
+#define FSL_SAI_TDR0	0x20 /* SAI Transmit Data 0 */
+#define FSL_SAI_TDR1	0x24 /* SAI Transmit Data 1 */
+#define FSL_SAI_TDR2	0x28 /* SAI Transmit Data 2 */
+#define FSL_SAI_TDR3	0x2C /* SAI Transmit Data 3 */
+#define FSL_SAI_TDR4	0x30 /* SAI Transmit Data 4 */
+#define FSL_SAI_TDR5	0x34 /* SAI Transmit Data 5 */
+#define FSL_SAI_TDR6	0x38 /* SAI Transmit Data 6 */
+#define FSL_SAI_TDR7	0x3C /* SAI Transmit Data 7 */
+#define FSL_SAI_TFR0	0x40 /* SAI Transmit FIFO 0 */
+#define FSL_SAI_TFR1	0x44 /* SAI Transmit FIFO 1 */
+#define FSL_SAI_TFR2	0x48 /* SAI Transmit FIFO 2 */
+#define FSL_SAI_TFR3	0x4C /* SAI Transmit FIFO 3 */
+#define FSL_SAI_TFR4	0x50 /* SAI Transmit FIFO 4 */
+#define FSL_SAI_TFR5	0x54 /* SAI Transmit FIFO 5 */
+#define FSL_SAI_TFR6	0x58 /* SAI Transmit FIFO 6 */
+#define FSL_SAI_TFR7	0x5C /* SAI Transmit FIFO 7 */
 #define FSL_SAI_TMR	0x60 /* SAI Transmit Mask */
 #define FSL_SAI_RCSR	0x80 /* SAI Receive Control */
 #define FSL_SAI_RCR1	0x84 /* SAI Receive Configuration 1 */
@@ -29,8 +43,22 @@
 #define FSL_SAI_RCR3	0x8c /* SAI Receive Configuration 3 */
 #define FSL_SAI_RCR4	0x90 /* SAI Receive Configuration 4 */
 #define FSL_SAI_RCR5	0x94 /* SAI Receive Configuration 5 */
-#define FSL_SAI_RDR	0xa0 /* SAI Receive Data */
-#define FSL_SAI_RFR	0xc0 /* SAI Receive FIFO */
+#define FSL_SAI_RDR0	0xa0 /* SAI Receive Data 0 */
+#define FSL_SAI_RDR1	0xa4 /* SAI Receive Data 1 */
+#define FSL_SAI_RDR2	0xa8 /* SAI Receive Data 2 */
+#define FSL_SAI_RDR3	0xac /* SAI Receive Data 3 */
+#define FSL_SAI_RDR4	0xb0 /* SAI Receive Data 4 */
+#define FSL_SAI_RDR5	0xb4 /* SAI Receive Data 5 */
+#define FSL_SAI_RDR6	0xb8 /* SAI Receive Data 6 */
+#define FSL_SAI_RDR7	0xbc /* SAI Receive Data 7 */
+#define FSL_SAI_RFR0	0xc0 /* SAI Receive FIFO 0 */
+#define FSL_SAI_RFR1	0xc4 /* SAI Receive FIFO 1 */
+#define FSL_SAI_RFR2	0xc8 /* SAI Receive FIFO 2 */
+#define FSL_SAI_RFR3	0xcc /* SAI Receive FIFO 3 */
+#define FSL_SAI_RFR4	0xd0 /* SAI Receive FIFO 4 */
+#define FSL_SAI_RFR5	0xd4 /* SAI Receive FIFO 5 */
+#define FSL_SAI_RFR6	0xd8 /* SAI Receive FIFO 6 */
+#define FSL_SAI_RFR7	0xdc /* SAI Receive FIFO 7 */
 #define FSL_SAI_RMR	0xe0 /* SAI Receive Mask */
 
 #define FSL_SAI_xCSR(tx)	(tx ? FSL_SAI_TCSR : FSL_SAI_RCSR)
-- 
cgit v1.2.3-59-g8ed1b


From b84f50b0fcb497a62068926fca793d2d213c7dbd Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Tue, 6 Aug 2019 18:12:11 +0300
Subject: ASoC: fsl_sai: Update Tx/Rx channel enable mask

Tx channel enable (TCE) / Rx channel enable (RCE) bits
enable corresponding data channel for Tx/Rx operation.

Because SAI supports up the 8 channels TCE/RCE occupy
up the 8 bits inside TCR3/RCR3 registers we need to extend
the mask to reflect this.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20190806151214.6783-3-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 6 ++++--
 sound/soc/fsl/fsl_sai.h | 1 +
 2 files changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index e4221f2a5ee3..f2698c94c9fe 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -599,7 +599,8 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	int ret;
 
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE,
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx),
+			   FSL_SAI_CR3_TRCE_MASK,
 			   FSL_SAI_CR3_TRCE);
 
 	ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
@@ -614,7 +615,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, 0);
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx),
+			   FSL_SAI_CR3_TRCE_MASK, 0);
 }
 
 static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 4bb478041d67..20c5b9b1e8bc 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -110,6 +110,7 @@
 
 /* SAI Transmit and Receive Configuration 3 Register */
 #define FSL_SAI_CR3_TRCE	BIT(16)
+#define FSL_SAI_CR3_TRCE_MASK	GENMASK(23, 16)
 #define FSL_SAI_CR3_WDFL(x)	(x)
 #define FSL_SAI_CR3_WDFL_MASK	0x1f
 
-- 
cgit v1.2.3-59-g8ed1b


From 4f7a0728b5305e2d865f543fbcffd617e03c7674 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Tue, 6 Aug 2019 18:12:12 +0300
Subject: ASoC: fsl_sai: Add support for SAI new version

New IP version introduces Version ID and Parameter registers
and optionally added Timestamp feature.

VERID and PARAM registers are placed at the top of registers
address space and some registers are shifted according to
the following table:

Tx/Rx data registers and Tx/Rx FIFO registers keep their
addresses, all other registers are shifted by 8.

SAI Memory map is described in chapter 13.10.4.1.1 I2S Memory map
of the Reference Manual [1].

In order to make as less changes as possible we attach an offset
to each register offset to each changed register definition. The
offset is read from each board private data.

[1]https://cache.nxp.com/secured/assets/documents/en/reference-manual/IMX8MDQLQRM.pdf?__gda__=1563728701_38bea7f0f726472cc675cb141b91bec7&fileExt=.pdf

Signed-off-by: Mihai Serban <mihai.serban@nxp.com>
[initial coding in the NXP internal tree]
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
[bugfixing and cleanups]
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
[adapted to linux-next]
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20190806151214.6783-4-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 228 ++++++++++++++++++++++++++++--------------------
 sound/soc/fsl/fsl_sai.h |  41 ++++-----
 2 files changed, 156 insertions(+), 113 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index f2698c94c9fe..0c5452927c04 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -40,6 +40,7 @@ static const struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = {
 static irqreturn_t fsl_sai_isr(int irq, void *devid)
 {
 	struct fsl_sai *sai = (struct fsl_sai *)devid;
+	unsigned int ofs = sai->soc_data->reg_offset;
 	struct device *dev = &sai->pdev->dev;
 	u32 flags, xcsr, mask;
 	bool irq_none = true;
@@ -52,7 +53,7 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid)
 	mask = (FSL_SAI_FLAGS >> FSL_SAI_CSR_xIE_SHIFT) << FSL_SAI_CSR_xF_SHIFT;
 
 	/* Tx IRQ */
-	regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr);
+	regmap_read(sai->regmap, FSL_SAI_TCSR(ofs), &xcsr);
 	flags = xcsr & mask;
 
 	if (flags)
@@ -82,11 +83,11 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid)
 	xcsr &= ~FSL_SAI_CSR_xF_MASK;
 
 	if (flags)
-		regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr);
+		regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), flags | xcsr);
 
 irq_rx:
 	/* Rx IRQ */
-	regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr);
+	regmap_read(sai->regmap, FSL_SAI_RCSR(ofs), &xcsr);
 	flags = xcsr & mask;
 
 	if (flags)
@@ -116,7 +117,7 @@ irq_rx:
 	xcsr &= ~FSL_SAI_CSR_xF_MASK;
 
 	if (flags)
-		regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr);
+		regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), flags | xcsr);
 
 out:
 	if (irq_none)
@@ -140,6 +141,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
 		int clk_id, unsigned int freq, int fsl_dir)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = fsl_dir == FSL_FMT_TRANSMITTER;
 	u32 val_cr2 = 0;
 
@@ -160,7 +162,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
 		return -EINVAL;
 	}
 
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx),
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs),
 			   FSL_SAI_CR2_MSEL_MASK, val_cr2);
 
 	return 0;
@@ -193,6 +195,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 				unsigned int fmt, int fsl_dir)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = fsl_dir == FSL_FMT_TRANSMITTER;
 	u32 val_cr2 = 0, val_cr4 = 0;
 
@@ -287,9 +290,9 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 		return -EINVAL;
 	}
 
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx),
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs),
 			   FSL_SAI_CR2_BCP | FSL_SAI_CR2_BCD_MSTR, val_cr2);
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx),
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
 			   FSL_SAI_CR4_MF | FSL_SAI_CR4_FSE |
 			   FSL_SAI_CR4_FSP | FSL_SAI_CR4_FSD_MSTR, val_cr4);
 
@@ -316,6 +319,7 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai);
+	unsigned int ofs = sai->soc_data->reg_offset;
 	unsigned long clk_rate;
 	u32 savediv = 0, ratio, savesub = freq;
 	u32 id;
@@ -378,17 +382,17 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
 	 */
 	if ((sai->synchronous[TX] && !sai->synchronous[RX]) ||
 	    (!tx && !sai->synchronous[RX])) {
-		regmap_update_bits(sai->regmap, FSL_SAI_RCR2,
+		regmap_update_bits(sai->regmap, FSL_SAI_RCR2(ofs),
 				   FSL_SAI_CR2_MSEL_MASK,
 				   FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
-		regmap_update_bits(sai->regmap, FSL_SAI_RCR2,
+		regmap_update_bits(sai->regmap, FSL_SAI_RCR2(ofs),
 				   FSL_SAI_CR2_DIV_MASK, savediv - 1);
 	} else if ((sai->synchronous[RX] && !sai->synchronous[TX]) ||
 		   (tx && !sai->synchronous[TX])) {
-		regmap_update_bits(sai->regmap, FSL_SAI_TCR2,
+		regmap_update_bits(sai->regmap, FSL_SAI_TCR2(ofs),
 				   FSL_SAI_CR2_MSEL_MASK,
 				   FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
-		regmap_update_bits(sai->regmap, FSL_SAI_TCR2,
+		regmap_update_bits(sai->regmap, FSL_SAI_TCR2(ofs),
 				   FSL_SAI_CR2_DIV_MASK, savediv - 1);
 	}
 
@@ -403,6 +407,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	unsigned int channels = params_channels(params);
 	u32 word_width = params_width(params);
@@ -455,19 +460,19 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 
 	if (!sai->is_slave_mode) {
 		if (!sai->synchronous[TX] && sai->synchronous[RX] && !tx) {
-			regmap_update_bits(sai->regmap, FSL_SAI_TCR4,
+			regmap_update_bits(sai->regmap, FSL_SAI_TCR4(ofs),
 				FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK,
 				val_cr4);
-			regmap_update_bits(sai->regmap, FSL_SAI_TCR5,
+			regmap_update_bits(sai->regmap, FSL_SAI_TCR5(ofs),
 				FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK |
 				FSL_SAI_CR5_FBT_MASK, val_cr5);
 			regmap_write(sai->regmap, FSL_SAI_TMR,
 				~0UL - ((1 << channels) - 1));
 		} else if (!sai->synchronous[RX] && sai->synchronous[TX] && tx) {
-			regmap_update_bits(sai->regmap, FSL_SAI_RCR4,
+			regmap_update_bits(sai->regmap, FSL_SAI_RCR4(ofs),
 				FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK,
 				val_cr4);
-			regmap_update_bits(sai->regmap, FSL_SAI_RCR5,
+			regmap_update_bits(sai->regmap, FSL_SAI_RCR5(ofs),
 				FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK |
 				FSL_SAI_CR5_FBT_MASK, val_cr5);
 			regmap_write(sai->regmap, FSL_SAI_RMR,
@@ -475,10 +480,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx),
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
 			   FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK,
 			   val_cr4);
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx),
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs),
 			   FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK |
 			   FSL_SAI_CR5_FBT_MASK, val_cr5);
 	regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << channels) - 1));
@@ -506,6 +511,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 		struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int ofs = sai->soc_data->reg_offset;
+
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 xcsr, count = 100;
 
@@ -514,9 +521,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 	 * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
 	 * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
 	 */
-	regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
-		           sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0);
-	regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
+	regmap_update_bits(sai->regmap, FSL_SAI_TCR2(ofs), FSL_SAI_CR2_SYNC,
+			   sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0);
+	regmap_update_bits(sai->regmap, FSL_SAI_RCR2(ofs), FSL_SAI_CR2_SYNC,
 			   sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
 
 	/*
@@ -527,43 +534,44 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs),
 				   FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE);
 
-		regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+		regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs),
 				   FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
-		regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+		regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs),
 				   FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
 
-		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs),
 				   FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs),
 				   FSL_SAI_CSR_FRDE, 0);
-		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+		regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs),
 				   FSL_SAI_CSR_xIE_MASK, 0);
 
 		/* Check if the opposite FRDE is also disabled */
-		regmap_read(sai->regmap, FSL_SAI_xCSR(!tx), &xcsr);
+		regmap_read(sai->regmap, FSL_SAI_xCSR(!tx, ofs), &xcsr);
 		if (!(xcsr & FSL_SAI_CSR_FRDE)) {
 			/* Disable both directions and reset their FIFOs */
-			regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+			regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs),
 					   FSL_SAI_CSR_TERE, 0);
-			regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+			regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs),
 					   FSL_SAI_CSR_TERE, 0);
 
 			/* TERE will remain set till the end of current frame */
 			do {
 				udelay(10);
-				regmap_read(sai->regmap, FSL_SAI_xCSR(tx), &xcsr);
+				regmap_read(sai->regmap,
+					    FSL_SAI_xCSR(tx, ofs), &xcsr);
 			} while (--count && xcsr & FSL_SAI_CSR_TERE);
 
-			regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+			regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs),
 					   FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
-			regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+			regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs),
 					   FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
 
 			/*
@@ -575,13 +583,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 			 */
 			if (!sai->is_slave_mode) {
 				/* Software Reset for both Tx and Rx */
-				regmap_write(sai->regmap,
-					     FSL_SAI_TCSR, FSL_SAI_CSR_SR);
-				regmap_write(sai->regmap,
-					     FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+				regmap_write(sai->regmap, FSL_SAI_TCSR(ofs),
+					     FSL_SAI_CSR_SR);
+				regmap_write(sai->regmap, FSL_SAI_RCSR(ofs),
+					     FSL_SAI_CSR_SR);
 				/* Clear SR bit to finish the reset */
-				regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
-				regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+				regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0);
+				regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
 			}
 		}
 		break;
@@ -596,10 +604,11 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	int ret;
 
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx),
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs),
 			   FSL_SAI_CR3_TRCE_MASK,
 			   FSL_SAI_CR3_TRCE);
 
@@ -613,9 +622,10 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
-	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx),
+	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs),
 			   FSL_SAI_CR3_TRCE_MASK, 0);
 }
 
@@ -633,18 +643,20 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
 static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
+	unsigned int ofs = sai->soc_data->reg_offset;
 
 	/* Software Reset for both Tx and Rx */
-	regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
-	regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+	regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR);
+	regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR);
 	/* Clear SR bit to finish the reset */
-	regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
-	regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+	regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0);
+	regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
 
-	regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
+	regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs),
+			   FSL_SAI_CR1_RFW_MASK,
 			   sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX);
-	regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
-			   FSL_SAI_MAXBURST_RX - 1);
+	regmap_update_bits(sai->regmap, FSL_SAI_RCR1(ofs),
+			   FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_RX - 1);
 
 	snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
 				&sai->dma_params_rx);
@@ -681,12 +693,12 @@ static const struct snd_soc_component_driver fsl_component = {
 	.name           = "fsl-sai",
 };
 
-static struct reg_default fsl_sai_reg_defaults[] = {
-	{FSL_SAI_TCR1, 0},
-	{FSL_SAI_TCR2, 0},
-	{FSL_SAI_TCR3, 0},
-	{FSL_SAI_TCR4, 0},
-	{FSL_SAI_TCR5, 0},
+static struct reg_default fsl_sai_reg_defaults_ofs0[] = {
+	{FSL_SAI_TCR1(0), 0},
+	{FSL_SAI_TCR2(0), 0},
+	{FSL_SAI_TCR3(0), 0},
+	{FSL_SAI_TCR4(0), 0},
+	{FSL_SAI_TCR5(0), 0},
 	{FSL_SAI_TDR0, 0},
 	{FSL_SAI_TDR1, 0},
 	{FSL_SAI_TDR2, 0},
@@ -695,24 +707,50 @@ static struct reg_default fsl_sai_reg_defaults[] = {
 	{FSL_SAI_TDR5, 0},
 	{FSL_SAI_TDR6, 0},
 	{FSL_SAI_TDR7, 0},
-	{FSL_SAI_TMR,  0},
-	{FSL_SAI_RCR1, 0},
-	{FSL_SAI_RCR2, 0},
-	{FSL_SAI_RCR3, 0},
-	{FSL_SAI_RCR4, 0},
-	{FSL_SAI_RCR5, 0},
-	{FSL_SAI_RMR,  0},
+	{FSL_SAI_TMR, 0},
+	{FSL_SAI_RCR1(0), 0},
+	{FSL_SAI_RCR2(0), 0},
+	{FSL_SAI_RCR3(0), 0},
+	{FSL_SAI_RCR4(0), 0},
+	{FSL_SAI_RCR5(0), 0},
+	{FSL_SAI_RMR, 0},
+};
+
+static struct reg_default fsl_sai_reg_defaults_ofs8[] = {
+	{FSL_SAI_TCR1(8), 0},
+	{FSL_SAI_TCR2(8), 0},
+	{FSL_SAI_TCR3(8), 0},
+	{FSL_SAI_TCR4(8), 0},
+	{FSL_SAI_TCR5(8), 0},
+	{FSL_SAI_TDR0, 0},
+	{FSL_SAI_TDR1, 0},
+	{FSL_SAI_TDR2, 0},
+	{FSL_SAI_TDR3, 0},
+	{FSL_SAI_TDR4, 0},
+	{FSL_SAI_TDR5, 0},
+	{FSL_SAI_TDR6, 0},
+	{FSL_SAI_TDR7, 0},
+	{FSL_SAI_TMR, 0},
+	{FSL_SAI_RCR1(8), 0},
+	{FSL_SAI_RCR2(8), 0},
+	{FSL_SAI_RCR3(8), 0},
+	{FSL_SAI_RCR4(8), 0},
+	{FSL_SAI_RCR5(8), 0},
+	{FSL_SAI_RMR, 0},
 };
 
 static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
 {
+	struct fsl_sai *sai = dev_get_drvdata(dev);
+	unsigned int ofs = sai->soc_data->reg_offset;
+
+	if (reg >= FSL_SAI_TCSR(ofs) && reg <= FSL_SAI_TCR5(ofs))
+		return true;
+
+	if (reg >= FSL_SAI_RCSR(ofs) && reg <= FSL_SAI_RCR5(ofs))
+		return true;
+
 	switch (reg) {
-	case FSL_SAI_TCSR:
-	case FSL_SAI_TCR1:
-	case FSL_SAI_TCR2:
-	case FSL_SAI_TCR3:
-	case FSL_SAI_TCR4:
-	case FSL_SAI_TCR5:
 	case FSL_SAI_TFR0:
 	case FSL_SAI_TFR1:
 	case FSL_SAI_TFR2:
@@ -722,12 +760,6 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
 	case FSL_SAI_TFR6:
 	case FSL_SAI_TFR7:
 	case FSL_SAI_TMR:
-	case FSL_SAI_RCSR:
-	case FSL_SAI_RCR1:
-	case FSL_SAI_RCR2:
-	case FSL_SAI_RCR3:
-	case FSL_SAI_RCR4:
-	case FSL_SAI_RCR5:
 	case FSL_SAI_RDR0:
 	case FSL_SAI_RDR1:
 	case FSL_SAI_RDR2:
@@ -753,9 +785,13 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
 
 static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
 {
+	struct fsl_sai *sai = dev_get_drvdata(dev);
+	unsigned int ofs = sai->soc_data->reg_offset;
+
+	if (reg == FSL_SAI_TCSR(ofs) || reg == FSL_SAI_RCSR(ofs))
+		return true;
+
 	switch (reg) {
-	case FSL_SAI_TCSR:
-	case FSL_SAI_RCSR:
 	case FSL_SAI_TFR0:
 	case FSL_SAI_TFR1:
 	case FSL_SAI_TFR2:
@@ -788,13 +824,16 @@ static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
 
 static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 {
+	struct fsl_sai *sai = dev_get_drvdata(dev);
+	unsigned int ofs = sai->soc_data->reg_offset;
+
+	if (reg >= FSL_SAI_TCSR(ofs) && reg <= FSL_SAI_TCR5(ofs))
+		return true;
+
+	if (reg >= FSL_SAI_RCSR(ofs) && reg <= FSL_SAI_RCR5(ofs))
+		return true;
+
 	switch (reg) {
-	case FSL_SAI_TCSR:
-	case FSL_SAI_TCR1:
-	case FSL_SAI_TCR2:
-	case FSL_SAI_TCR3:
-	case FSL_SAI_TCR4:
-	case FSL_SAI_TCR5:
 	case FSL_SAI_TDR0:
 	case FSL_SAI_TDR1:
 	case FSL_SAI_TDR2:
@@ -804,12 +843,6 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 	case FSL_SAI_TDR6:
 	case FSL_SAI_TDR7:
 	case FSL_SAI_TMR:
-	case FSL_SAI_RCSR:
-	case FSL_SAI_RCR1:
-	case FSL_SAI_RCR2:
-	case FSL_SAI_RCR3:
-	case FSL_SAI_RCR4:
-	case FSL_SAI_RCR5:
 	case FSL_SAI_RMR:
 		return true;
 	default:
@@ -817,15 +850,15 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static const struct regmap_config fsl_sai_regmap_config = {
+static struct regmap_config fsl_sai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
 	.fast_io = true,
 
 	.max_register = FSL_SAI_RMR,
-	.reg_defaults = fsl_sai_reg_defaults,
-	.num_reg_defaults = ARRAY_SIZE(fsl_sai_reg_defaults),
+	.reg_defaults = fsl_sai_reg_defaults_ofs0,
+	.num_reg_defaults = ARRAY_SIZE(fsl_sai_reg_defaults_ofs0),
 	.readable_reg = fsl_sai_readable_reg,
 	.volatile_reg = fsl_sai_volatile_reg,
 	.writeable_reg = fsl_sai_writeable_reg,
@@ -857,6 +890,12 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
+	if (sai->soc_data->reg_offset == 8) {
+		fsl_sai_regmap_config.reg_defaults = fsl_sai_reg_defaults_ofs8;
+		fsl_sai_regmap_config.num_reg_defaults =
+			ARRAY_SIZE(fsl_sai_reg_defaults_ofs8);
+	}
+
 	sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
 			"bus", base, &fsl_sai_regmap_config);
 
@@ -971,11 +1010,13 @@ static int fsl_sai_remove(struct platform_device *pdev)
 static const struct fsl_sai_soc_data fsl_sai_vf610_data = {
 	.use_imx_pcm = false,
 	.fifo_depth = 32,
+	.reg_offset = 0,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
 	.use_imx_pcm = true,
 	.fifo_depth = 32,
+	.reg_offset = 0,
 };
 
 static const struct of_device_id fsl_sai_ids[] = {
@@ -1008,6 +1049,7 @@ static int fsl_sai_runtime_suspend(struct device *dev)
 static int fsl_sai_runtime_resume(struct device *dev)
 {
 	struct fsl_sai *sai = dev_get_drvdata(dev);
+	unsigned int ofs = sai->soc_data->reg_offset;
 	int ret;
 
 	ret = clk_prepare_enable(sai->bus_clk);
@@ -1029,11 +1071,11 @@ static int fsl_sai_runtime_resume(struct device *dev)
 	}
 
 	regcache_cache_only(sai->regmap, false);
-	regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
-	regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+	regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR);
+	regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR);
 	usleep_range(1000, 2000);
-	regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
-	regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+	regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0);
+	regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
 
 	ret = regcache_sync(sai->regmap);
 	if (ret)
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 20c5b9b1e8bc..b89b0ca26053 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -14,12 +14,12 @@
 			 SNDRV_PCM_FMTBIT_S32_LE)
 
 /* SAI Register Map Register */
-#define FSL_SAI_TCSR	0x00 /* SAI Transmit Control */
-#define FSL_SAI_TCR1	0x04 /* SAI Transmit Configuration 1 */
-#define FSL_SAI_TCR2	0x08 /* SAI Transmit Configuration 2 */
-#define FSL_SAI_TCR3	0x0c /* SAI Transmit Configuration 3 */
-#define FSL_SAI_TCR4	0x10 /* SAI Transmit Configuration 4 */
-#define FSL_SAI_TCR5	0x14 /* SAI Transmit Configuration 5 */
+#define FSL_SAI_TCSR(ofs)	(0x00 + ofs) /* SAI Transmit Control */
+#define FSL_SAI_TCR1(ofs)	(0x04 + ofs) /* SAI Transmit Configuration 1 */
+#define FSL_SAI_TCR2(ofs)	(0x08 + ofs) /* SAI Transmit Configuration 2 */
+#define FSL_SAI_TCR3(ofs)	(0x0c + ofs) /* SAI Transmit Configuration 3 */
+#define FSL_SAI_TCR4(ofs)	(0x10 + ofs) /* SAI Transmit Configuration 4 */
+#define FSL_SAI_TCR5(ofs)	(0x14 + ofs) /* SAI Transmit Configuration 5 */
 #define FSL_SAI_TDR0	0x20 /* SAI Transmit Data 0 */
 #define FSL_SAI_TDR1	0x24 /* SAI Transmit Data 1 */
 #define FSL_SAI_TDR2	0x28 /* SAI Transmit Data 2 */
@@ -37,12 +37,12 @@
 #define FSL_SAI_TFR6	0x58 /* SAI Transmit FIFO 6 */
 #define FSL_SAI_TFR7	0x5C /* SAI Transmit FIFO 7 */
 #define FSL_SAI_TMR	0x60 /* SAI Transmit Mask */
-#define FSL_SAI_RCSR	0x80 /* SAI Receive Control */
-#define FSL_SAI_RCR1	0x84 /* SAI Receive Configuration 1 */
-#define FSL_SAI_RCR2	0x88 /* SAI Receive Configuration 2 */
-#define FSL_SAI_RCR3	0x8c /* SAI Receive Configuration 3 */
-#define FSL_SAI_RCR4	0x90 /* SAI Receive Configuration 4 */
-#define FSL_SAI_RCR5	0x94 /* SAI Receive Configuration 5 */
+#define FSL_SAI_RCSR(ofs)	(0x80 + ofs) /* SAI Receive Control */
+#define FSL_SAI_RCR1(ofs)	(0x84 + ofs)/* SAI Receive Configuration 1 */
+#define FSL_SAI_RCR2(ofs)	(0x88 + ofs) /* SAI Receive Configuration 2 */
+#define FSL_SAI_RCR3(ofs)	(0x8c + ofs) /* SAI Receive Configuration 3 */
+#define FSL_SAI_RCR4(ofs)	(0x90 + ofs) /* SAI Receive Configuration 4 */
+#define FSL_SAI_RCR5(ofs)	(0x94 + ofs) /* SAI Receive Configuration 5 */
 #define FSL_SAI_RDR0	0xa0 /* SAI Receive Data 0 */
 #define FSL_SAI_RDR1	0xa4 /* SAI Receive Data 1 */
 #define FSL_SAI_RDR2	0xa8 /* SAI Receive Data 2 */
@@ -61,14 +61,14 @@
 #define FSL_SAI_RFR7	0xdc /* SAI Receive FIFO 7 */
 #define FSL_SAI_RMR	0xe0 /* SAI Receive Mask */
 
-#define FSL_SAI_xCSR(tx)	(tx ? FSL_SAI_TCSR : FSL_SAI_RCSR)
-#define FSL_SAI_xCR1(tx)	(tx ? FSL_SAI_TCR1 : FSL_SAI_RCR1)
-#define FSL_SAI_xCR2(tx)	(tx ? FSL_SAI_TCR2 : FSL_SAI_RCR2)
-#define FSL_SAI_xCR3(tx)	(tx ? FSL_SAI_TCR3 : FSL_SAI_RCR3)
-#define FSL_SAI_xCR4(tx)	(tx ? FSL_SAI_TCR4 : FSL_SAI_RCR4)
-#define FSL_SAI_xCR5(tx)	(tx ? FSL_SAI_TCR5 : FSL_SAI_RCR5)
-#define FSL_SAI_xDR(tx)		(tx ? FSL_SAI_TDR : FSL_SAI_RDR)
-#define FSL_SAI_xFR(tx)		(tx ? FSL_SAI_TFR : FSL_SAI_RFR)
+#define FSL_SAI_xCSR(tx, ofs)	(tx ? FSL_SAI_TCSR(ofs) : FSL_SAI_RCSR(ofs))
+#define FSL_SAI_xCR1(tx, ofs)	(tx ? FSL_SAI_TCR1(ofs) : FSL_SAI_RCR1(ofs))
+#define FSL_SAI_xCR2(tx, ofs)	(tx ? FSL_SAI_TCR2(ofs) : FSL_SAI_RCR2(ofs))
+#define FSL_SAI_xCR3(tx, ofs)	(tx ? FSL_SAI_TCR3(ofs) : FSL_SAI_RCR3(ofs))
+#define FSL_SAI_xCR4(tx, ofs)	(tx ? FSL_SAI_TCR4(ofs) : FSL_SAI_RCR4(ofs))
+#define FSL_SAI_xCR5(tx, ofs)	(tx ? FSL_SAI_TCR5(ofs) : FSL_SAI_RCR5(ofs))
+#define FSL_SAI_xDR(tx, ofs)	(tx ? FSL_SAI_TDR(ofs) : FSL_SAI_RDR(ofs))
+#define FSL_SAI_xFR(tx, ofs)	(tx ? FSL_SAI_TFR(ofs) : FSL_SAI_RFR(ofs))
 #define FSL_SAI_xMR(tx)		(tx ? FSL_SAI_TMR : FSL_SAI_RMR)
 
 /* SAI Transmit/Receive Control Register */
@@ -158,6 +158,7 @@
 struct fsl_sai_soc_data {
 	bool use_imx_pcm;
 	unsigned int fifo_depth;
+	unsigned int reg_offset;
 };
 
 struct fsl_sai {
-- 
cgit v1.2.3-59-g8ed1b


From a860fac420971c5a90d4f78959b44ead793aee4f Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Tue, 6 Aug 2019 18:12:13 +0300
Subject: ASoC: fsl_sai: Add support for imx7ulp/imx8mq

SAI module on imx7ulp/imx8m features 2 new registers (VERID and PARAM)
at the beginning of register address space.

On imx7ulp FIFOs can held up to 16 x 32 bit samples.
On imx8mq FIFOs can held up to 128 x 32 bit samples.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20190806151214.6783-5-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 0c5452927c04..4a346fcb5630 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1019,10 +1019,24 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
 	.reg_offset = 0,
 };
 
+static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = {
+	.use_imx_pcm = true,
+	.fifo_depth = 16,
+	.reg_offset = 8,
+};
+
+static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = {
+	.use_imx_pcm = true,
+	.fifo_depth = 128,
+	.reg_offset = 8,
+};
+
 static const struct of_device_id fsl_sai_ids[] = {
 	{ .compatible = "fsl,vf610-sai", .data = &fsl_sai_vf610_data },
 	{ .compatible = "fsl,imx6sx-sai", .data = &fsl_sai_imx6sx_data },
 	{ .compatible = "fsl,imx6ul-sai", .data = &fsl_sai_imx6sx_data },
+	{ .compatible = "fsl,imx7ulp-sai", .data = &fsl_sai_imx7ulp_data },
+	{ .compatible = "fsl,imx8mq-sai", .data = &fsl_sai_imx8mq_data },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(of, fsl_sai_ids);
-- 
cgit v1.2.3-59-g8ed1b


From 619a1f195f93276dc8c6e33fe057e007adc9c288 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 7 Aug 2019 20:02:31 +0200
Subject: ALSA: hda: Remove page allocation redirection

The HD-audio core allocates and releases pages via driver's specific
dma_alloc_pages and dma_free_pages ops defined in bus->io_ops.  This
was because some platforms require the uncached pages and the handling
of page flags had to be done locally in the driver code.

Since the recent change in ALSA core memory allocator, we can simply
pass SNDRV_DMA_TYPE_DEV_UC for the uncached pages, and the only
difference became about this type to be passed to the core allocator.
That is, it's good time for cleaning up the mess.

This patch changes the allocation code in HD-audio core to call the
core allocator directly so that we get rid of dma_alloc_pages and
dma_free_pages io_ops.  If a driver needs the uncached pages, it has
to set bus->dma_type right after the bus initialization.

This is merely a code refactoring and shouldn't bring any behavior
changes.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hdaudio.h                |  6 +-----
 sound/hda/ext/hdac_ext_bus.c           | 13 -------------
 sound/hda/hdac_bus.c                   |  1 +
 sound/hda/hdac_controller.c            | 18 +++++++++---------
 sound/hda/hdac_stream.c                |  8 ++++----
 sound/pci/hda/hda_intel.c              | 24 ++++--------------------
 sound/pci/hda/hda_tegra.c              | 16 ----------------
 sound/soc/intel/skylake/skl-messages.c | 15 ++-------------
 sound/soc/sof/intel/hda-bus.c          | 14 --------------
 9 files changed, 21 insertions(+), 94 deletions(-)

(limited to 'sound')

diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 612a17e375d0..20549def0a27 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -264,11 +264,6 @@ struct hdac_io_ops {
 	u16 (*reg_readw)(u16 __iomem *addr);
 	void (*reg_writeb)(u8 value, u8 __iomem *addr);
 	u8 (*reg_readb)(u8 __iomem *addr);
-	/* Allocation ops */
-	int (*dma_alloc_pages)(struct hdac_bus *bus, int type, size_t size,
-			       struct snd_dma_buffer *buf);
-	void (*dma_free_pages)(struct hdac_bus *bus,
-			       struct snd_dma_buffer *buf);
 };
 
 #define HDA_UNSOL_QUEUE_SIZE	64
@@ -344,6 +339,7 @@ struct hdac_bus {
 	/* CORB/RIRB and position buffers */
 	struct snd_dma_buffer rb;
 	struct snd_dma_buffer posbuf;
+	int dma_type;			/* SNDRV_DMA_TYPE_XXX for CORB/RIRB */
 
 	/* hdac_stream linked list */
 	struct list_head stream_list;
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 4f9f1d2a2ec5..7825b74068f4 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -47,17 +47,6 @@ static u8 hdac_ext_readb(u8 __iomem *addr)
 	return readb(addr);
 }
 
-static int hdac_ext_dma_alloc_pages(struct hdac_bus *bus, int type,
-			   size_t size, struct snd_dma_buffer *buf)
-{
-	return snd_dma_alloc_pages(type, bus->dev, size, buf);
-}
-
-static void hdac_ext_dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
-{
-	snd_dma_free_pages(buf);
-}
-
 static const struct hdac_io_ops hdac_ext_default_io = {
 	.reg_writel = hdac_ext_writel,
 	.reg_readl = hdac_ext_readl,
@@ -65,8 +54,6 @@ static const struct hdac_io_ops hdac_ext_default_io = {
 	.reg_readw = hdac_ext_readw,
 	.reg_writeb = hdac_ext_writeb,
 	.reg_readb = hdac_ext_readb,
-	.dma_alloc_pages = hdac_ext_dma_alloc_pages,
-	.dma_free_pages = hdac_ext_dma_free_pages,
 };
 
 /**
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 14e57ffd5bc1..00ea12e67dc8 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -34,6 +34,7 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
 	else
 		bus->ops = &default_ops;
 	bus->io_ops = io_ops;
+	bus->dma_type = SNDRV_DMA_TYPE_DEV;
 	INIT_LIST_HEAD(&bus->stream_list);
 	INIT_LIST_HEAD(&bus->codec_list);
 	INIT_WORK(&bus->unsol_work, snd_hdac_bus_process_unsol_events);
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 3b0110545070..7e7be8e4dcf9 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -575,12 +575,13 @@ int snd_hdac_bus_alloc_stream_pages(struct hdac_bus *bus)
 {
 	struct hdac_stream *s;
 	int num_streams = 0;
+	int dma_type = bus->dma_type ? bus->dma_type : SNDRV_DMA_TYPE_DEV;
 	int err;
 
 	list_for_each_entry(s, &bus->stream_list, list) {
 		/* allocate memory for the BDL for each stream */
-		err = bus->io_ops->dma_alloc_pages(bus, SNDRV_DMA_TYPE_DEV,
-						   BDL_SIZE, &s->bdl);
+		err = snd_dma_alloc_pages(dma_type, bus->dev,
+					  BDL_SIZE, &s->bdl);
 		num_streams++;
 		if (err < 0)
 			return -ENOMEM;
@@ -589,16 +590,15 @@ int snd_hdac_bus_alloc_stream_pages(struct hdac_bus *bus)
 	if (WARN_ON(!num_streams))
 		return -EINVAL;
 	/* allocate memory for the position buffer */
-	err = bus->io_ops->dma_alloc_pages(bus, SNDRV_DMA_TYPE_DEV,
-					   num_streams * 8, &bus->posbuf);
+	err = snd_dma_alloc_pages(dma_type, bus->dev,
+				  num_streams * 8, &bus->posbuf);
 	if (err < 0)
 		return -ENOMEM;
 	list_for_each_entry(s, &bus->stream_list, list)
 		s->posbuf = (__le32 *)(bus->posbuf.area + s->index * 8);
 
 	/* single page (at least 4096 bytes) must suffice for both ringbuffes */
-	return bus->io_ops->dma_alloc_pages(bus, SNDRV_DMA_TYPE_DEV,
-					    PAGE_SIZE, &bus->rb);
+	return snd_dma_alloc_pages(dma_type, bus->dev, PAGE_SIZE, &bus->rb);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_alloc_stream_pages);
 
@@ -612,12 +612,12 @@ void snd_hdac_bus_free_stream_pages(struct hdac_bus *bus)
 
 	list_for_each_entry(s, &bus->stream_list, list) {
 		if (s->bdl.area)
-			bus->io_ops->dma_free_pages(bus, &s->bdl);
+			snd_dma_free_pages(&s->bdl);
 	}
 
 	if (bus->rb.area)
-		bus->io_ops->dma_free_pages(bus, &bus->rb);
+		snd_dma_free_pages(&bus->rb);
 	if (bus->posbuf.area)
-		bus->io_ops->dma_free_pages(bus, &bus->posbuf);
+		snd_dma_free_pages(&bus->posbuf);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_free_stream_pages);
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index 55d53b89ac21..fc68d4ce0a37 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -680,8 +680,8 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format,
 	azx_dev->locked = true;
 	spin_unlock_irq(&bus->reg_lock);
 
-	err = bus->io_ops->dma_alloc_pages(bus, SNDRV_DMA_TYPE_DEV_SG,
-					   byte_size, bufp);
+	err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, bus->dev,
+				  byte_size, bufp);
 	if (err < 0)
 		goto err_alloc;
 
@@ -707,7 +707,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format,
 	return azx_dev->stream_tag;
 
  error:
-	bus->io_ops->dma_free_pages(bus, bufp);
+	snd_dma_free_pages(bufp);
  err_alloc:
 	spin_lock_irq(&bus->reg_lock);
 	azx_dev->locked = false;
@@ -754,7 +754,7 @@ void snd_hdac_dsp_cleanup(struct hdac_stream *azx_dev,
 	azx_dev->period_bytes = 0;
 	azx_dev->format_val = 0;
 
-	bus->io_ops->dma_free_pages(bus, dmab);
+	snd_dma_free_pages(dmab);
 	dmab->area = NULL;
 
 	spin_lock_irq(&bus->reg_lock);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cb8b0945547c..3bb4c26f2799 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1694,6 +1694,10 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
 		return err;
 	}
 
+	/* use the non-cached pages in non-snoop mode */
+	if (!azx_snoop(chip))
+		azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_UC;
+
 	/* Workaround for a communication error on CFL (bko#199007) and CNL */
 	if (IS_CFL(pci) || IS_CNL(pci))
 		azx_bus(chip)->polling_mode = 1;
@@ -1979,24 +1983,6 @@ static int disable_msi_reset_irq(struct azx *chip)
 	return 0;
 }
 
-/* DMA page allocation helpers.  */
-static int dma_alloc_pages(struct hdac_bus *bus,
-			   int type,
-			   size_t size,
-			   struct snd_dma_buffer *buf)
-{
-	struct azx *chip = bus_to_azx(bus);
-
-	if (!azx_snoop(chip) && type == SNDRV_DMA_TYPE_DEV)
-		type = SNDRV_DMA_TYPE_DEV_UC;
-	return snd_dma_alloc_pages(type, bus->dev, size, buf);
-}
-
-static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
-{
-	snd_dma_free_pages(buf);
-}
-
 static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
 			     struct vm_area_struct *area)
 {
@@ -2015,8 +2001,6 @@ static const struct hdac_io_ops pci_hda_io_ops = {
 	.reg_readw = pci_azx_readw,
 	.reg_writeb = pci_azx_writeb,
 	.reg_readb = pci_azx_readb,
-	.dma_alloc_pages = dma_alloc_pages,
-	.dma_free_pages = dma_free_pages,
 };
 
 static const struct hda_controller_ops pci_hda_ops = {
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 7dbe9f39fc79..ba414cc639f1 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -75,20 +75,6 @@ MODULE_PARM_DESC(power_save,
 #define power_save	0
 #endif
 
-/*
- * DMA page allocation ops.
- */
-static int dma_alloc_pages(struct hdac_bus *bus, int type, size_t size,
-			   struct snd_dma_buffer *buf)
-{
-	return snd_dma_alloc_pages(type, bus->dev, size, buf);
-}
-
-static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
-{
-	snd_dma_free_pages(buf);
-}
-
 /*
  * Register access ops. Tegra HDA register access is DWORD only.
  */
@@ -153,8 +139,6 @@ static const struct hdac_io_ops hda_tegra_io_ops = {
 	.reg_readw = hda_tegra_readw,
 	.reg_writeb = hda_tegra_writeb,
 	.reg_readb = hda_tegra_readb,
-	.dma_alloc_pages = dma_alloc_pages,
-	.dma_free_pages = dma_free_pages,
 };
 
 static const struct hda_controller_ops hda_tegra_ops; /* nothing special */
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index febc070839e0..c6f9e05c929e 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -25,23 +25,12 @@
 static int skl_alloc_dma_buf(struct device *dev,
 		struct snd_dma_buffer *dmab, size_t size)
 {
-	struct hdac_bus *bus = dev_get_drvdata(dev);
-
-	if (!bus)
-		return -ENODEV;
-
-	return  bus->io_ops->dma_alloc_pages(bus, SNDRV_DMA_TYPE_DEV, size, dmab);
+	return snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, dmab);
 }
 
 static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab)
 {
-	struct hdac_bus *bus = dev_get_drvdata(dev);
-
-	if (!bus)
-		return -ENODEV;
-
-	bus->io_ops->dma_free_pages(bus, dmab);
-
+	snd_dma_free_pages(dmab);
 	return 0;
 }
 
diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index a7e6d8227df6..0bc93fa06b5b 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -51,18 +51,6 @@ static u8 sof_hda_readb(u8 __iomem *addr)
 	return readb(addr);
 }
 
-static int sof_hda_dma_alloc_pages(struct hdac_bus *bus, int type,
-				   size_t size, struct snd_dma_buffer *buf)
-{
-	return snd_dma_alloc_pages(type, bus->dev, size, buf);
-}
-
-static void sof_hda_dma_free_pages(struct hdac_bus *bus,
-				   struct snd_dma_buffer *buf)
-{
-	snd_dma_free_pages(buf);
-}
-
 static const struct hdac_io_ops io_ops = {
 	.reg_writel = sof_hda_writel,
 	.reg_readl = sof_hda_readl,
@@ -70,8 +58,6 @@ static const struct hdac_io_ops io_ops = {
 	.reg_readw = sof_hda_readw,
 	.reg_writeb = sof_hda_writeb,
 	.reg_readb = sof_hda_readb,
-	.dma_alloc_pages = sof_hda_dma_alloc_pages,
-	.dma_free_pages = sof_hda_dma_free_pages,
 };
 
 /*
-- 
cgit v1.2.3-59-g8ed1b


From 19abfefd4c7604993d1c31e098a3f48bdafe334d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 7 Aug 2019 20:32:08 +0200
Subject: ALSA: hda: Direct MMIO accesses

HD-audio drivers access to the mmio registers indirectly via the
corresponding bus->io_ops callbacks.  This is because some platform
(notably Tegra SoC) requires the word-aligned access.  But it's rather
a rare case, and other platforms suffer from the penalties by indirect
calls unnecessarily.

This patch is an attempt to optimize and cleanup for this situation.
Now the special aligned access is used only when a new kconfig
CONFIG_SND_HDA_ALIGNED_MMIO is set.  And the HD-audio core itself
provides the aligned MMIO access helpers instead of the driver side.
If Kconfig isn't set (as default), the standard helpers like readl()
or writel() are used directly.

A couple of places in ASoC Intel drivers have the access via io_ops
reg_writel(), and they are replaced with the direct writel() calls.

And now with this patch, the whole bus->io_ops becomes empty, so it's
dropped completely.  The bus initialization functions are changed
accordingly as well to drop the whole bus->io_ops.

Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hdaudio.h        | 63 +++++++++++++++++++-------------------
 include/sound/hdaudio_ext.h    |  1 -
 sound/hda/Kconfig              |  3 ++
 sound/hda/ext/hdac_ext_bus.c   | 47 +----------------------------
 sound/hda/hdac_bus.c           | 35 +++++++++++++++++++---
 sound/pci/hda/Kconfig          |  1 +
 sound/pci/hda/hda_controller.c |  6 ++--
 sound/pci/hda/hda_controller.h |  3 +-
 sound/pci/hda/hda_intel.c      | 47 +----------------------------
 sound/pci/hda/hda_tegra.c      | 68 +-----------------------------------------
 sound/soc/intel/skylake/skl.c  |  7 ++---
 sound/soc/sof/intel/hda-bus.c  | 40 -------------------------
 sound/soc/sof/intel/hda-dsp.c  |  2 +-
 13 files changed, 75 insertions(+), 248 deletions(-)

(limited to 'sound')

diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 20549def0a27..4af4af55e854 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -253,19 +253,6 @@ struct hdac_ext_bus_ops {
 	int (*hdev_detach)(struct hdac_device *hdev);
 };
 
-/*
- * Lowlevel I/O operators
- */
-struct hdac_io_ops {
-	/* mapped register accesses */
-	void (*reg_writel)(u32 value, u32 __iomem *addr);
-	u32 (*reg_readl)(u32 __iomem *addr);
-	void (*reg_writew)(u16 value, u16 __iomem *addr);
-	u16 (*reg_readw)(u16 __iomem *addr);
-	void (*reg_writeb)(u8 value, u8 __iomem *addr);
-	u8 (*reg_readb)(u8 __iomem *addr);
-};
-
 #define HDA_UNSOL_QUEUE_SIZE	64
 #define HDA_MAX_CODECS		8	/* limit by controller side */
 
@@ -299,7 +286,6 @@ struct hdac_rb {
 struct hdac_bus {
 	struct device *dev;
 	const struct hdac_bus_ops *ops;
-	const struct hdac_io_ops *io_ops;
 	const struct hdac_ext_bus_ops *ext_ops;
 
 	/* h/w resources */
@@ -380,8 +366,7 @@ struct hdac_bus {
 };
 
 int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
-		      const struct hdac_bus_ops *ops,
-		      const struct hdac_io_ops *io_ops);
+		      const struct hdac_bus_ops *ops);
 void snd_hdac_bus_exit(struct hdac_bus *bus);
 int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr,
 			   unsigned int cmd, unsigned int *res);
@@ -425,21 +410,38 @@ int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
 int snd_hdac_bus_alloc_stream_pages(struct hdac_bus *bus);
 void snd_hdac_bus_free_stream_pages(struct hdac_bus *bus);
 
+#ifdef CONFIG_SND_HDA_ALIGNED_MMIO
+unsigned int snd_hdac_aligned_read(void __iomem *addr, unsigned int mask);
+void snd_hdac_aligned_write(unsigned int val, void __iomem *addr,
+			    unsigned int mask);
+#define snd_hdac_reg_writeb(v, addr)	snd_hdac_aligned_write(v, addr, 0xff)
+#define snd_hdac_reg_writew(v, addr)	snd_hdac_aligned_write(v, addr, 0xffff)
+#define snd_hdac_reg_readb(addr)	snd_hdac_aligned_read(addr, 0xff)
+#define snd_hdac_reg_readw(addr)	snd_hdac_aligned_read(addr, 0xffff)
+#else /* CONFIG_SND_HDA_ALIGNED_MMIO */
+#define snd_hdac_reg_writeb(val, addr)	writeb(val, addr)
+#define snd_hdac_reg_writew(val, addr)	writew(val, addr)
+#define snd_hdac_reg_readb(addr)	readb(addr)
+#define snd_hdac_reg_readw(addr)	readw(addr)
+#endif /* CONFIG_SND_HDA_ALIGNED_MMIO */
+#define snd_hdac_reg_writel(val, addr)	writel(val, addr)
+#define snd_hdac_reg_readl(addr)	readl(addr)
+
 /*
  * macros for easy use
  */
 #define _snd_hdac_chip_writeb(chip, reg, value) \
-	((chip)->io_ops->reg_writeb(value, (chip)->remap_addr + (reg)))
+	snd_hdac_reg_writeb(value, (chip)->remap_addr + (reg))
 #define _snd_hdac_chip_readb(chip, reg) \
-	((chip)->io_ops->reg_readb((chip)->remap_addr + (reg)))
+	snd_hdac_reg_readb((chip)->remap_addr + (reg))
 #define _snd_hdac_chip_writew(chip, reg, value) \
-	((chip)->io_ops->reg_writew(value, (chip)->remap_addr + (reg)))
+	snd_hdac_reg_writew(value, (chip)->remap_addr + (reg))
 #define _snd_hdac_chip_readw(chip, reg) \
-	((chip)->io_ops->reg_readw((chip)->remap_addr + (reg)))
+	snd_hdac_reg_readw((chip)->remap_addr + (reg))
 #define _snd_hdac_chip_writel(chip, reg, value) \
-	((chip)->io_ops->reg_writel(value, (chip)->remap_addr + (reg)))
+	snd_hdac_reg_writel(value, (chip)->remap_addr + (reg))
 #define _snd_hdac_chip_readl(chip, reg) \
-	((chip)->io_ops->reg_readl((chip)->remap_addr + (reg)))
+	snd_hdac_reg_readl((chip)->remap_addr + (reg))
 
 /* read/write a register, pass without AZX_REG_ prefix */
 #define snd_hdac_chip_writel(chip, reg, value) \
@@ -544,24 +546,19 @@ int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus,
 /*
  * macros for easy use
  */
-#define _snd_hdac_stream_write(type, dev, reg, value)			\
-	((dev)->bus->io_ops->reg_write ## type(value, (dev)->sd_addr + (reg)))
-#define _snd_hdac_stream_read(type, dev, reg)				\
-	((dev)->bus->io_ops->reg_read ## type((dev)->sd_addr + (reg)))
-
 /* read/write a register, pass without AZX_REG_ prefix */
 #define snd_hdac_stream_writel(dev, reg, value) \
-	_snd_hdac_stream_write(l, dev, AZX_REG_ ## reg, value)
+	snd_hdac_reg_writel(value, (dev)->sd_addr + AZX_REG_ ## reg)
 #define snd_hdac_stream_writew(dev, reg, value) \
-	_snd_hdac_stream_write(w, dev, AZX_REG_ ## reg, value)
+	snd_hdac_reg_writew(value, (dev)->sd_addr + AZX_REG_ ## reg)
 #define snd_hdac_stream_writeb(dev, reg, value) \
-	_snd_hdac_stream_write(b, dev, AZX_REG_ ## reg, value)
+	snd_hdac_reg_writeb(value, (dev)->sd_addr + AZX_REG_ ## reg)
 #define snd_hdac_stream_readl(dev, reg) \
-	_snd_hdac_stream_read(l, dev, AZX_REG_ ## reg)
+	snd_hdac_reg_readl((dev)->sd_addr + AZX_REG_ ## reg)
 #define snd_hdac_stream_readw(dev, reg) \
-	_snd_hdac_stream_read(w, dev, AZX_REG_ ## reg)
+	snd_hdac_reg_readw((dev)->sd_addr + AZX_REG_ ## reg)
 #define snd_hdac_stream_readb(dev, reg) \
-	_snd_hdac_stream_read(b, dev, AZX_REG_ ## reg)
+	snd_hdac_reg_readb((dev)->sd_addr + AZX_REG_ ## reg)
 
 /* update a register, pass without AZX_REG_ prefix */
 #define snd_hdac_stream_updatel(dev, reg, mask, val) \
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index f34aced69ca8..ef88b20c7b0a 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -6,7 +6,6 @@
 
 int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 		      const struct hdac_bus_ops *ops,
-		      const struct hdac_io_ops *io_ops,
 		      const struct hdac_ext_bus_ops *ext_ops);
 
 void snd_hdac_ext_bus_exit(struct hdac_bus *bus);
diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig
index f6feced15f17..1d475cf3f754 100644
--- a/sound/hda/Kconfig
+++ b/sound/hda/Kconfig
@@ -6,6 +6,9 @@ config SND_HDA_CORE
 config SND_HDA_DSP_LOADER
 	bool
 
+config SND_HDA_ALIGNED_MMIO
+	bool
+
 config SND_HDA_COMPONENT
 	bool
 
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 7825b74068f4..242306d820ec 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -17,67 +17,22 @@
 MODULE_DESCRIPTION("HDA extended core");
 MODULE_LICENSE("GPL v2");
 
-static void hdac_ext_writel(u32 value, u32 __iomem *addr)
-{
-	writel(value, addr);
-}
-
-static u32 hdac_ext_readl(u32 __iomem *addr)
-{
-	return readl(addr);
-}
-
-static void hdac_ext_writew(u16 value, u16 __iomem *addr)
-{
-	writew(value, addr);
-}
-
-static u16 hdac_ext_readw(u16 __iomem *addr)
-{
-	return readw(addr);
-}
-
-static void hdac_ext_writeb(u8 value, u8 __iomem *addr)
-{
-	writeb(value, addr);
-}
-
-static u8 hdac_ext_readb(u8 __iomem *addr)
-{
-	return readb(addr);
-}
-
-static const struct hdac_io_ops hdac_ext_default_io = {
-	.reg_writel = hdac_ext_writel,
-	.reg_readl = hdac_ext_readl,
-	.reg_writew = hdac_ext_writew,
-	.reg_readw = hdac_ext_readw,
-	.reg_writeb = hdac_ext_writeb,
-	.reg_readb = hdac_ext_readb,
-};
-
 /**
  * snd_hdac_ext_bus_init - initialize a HD-audio extended bus
  * @ebus: the pointer to extended bus object
  * @dev: device pointer
  * @ops: bus verb operators
- * @io_ops: lowlevel I/O operators, can be NULL. If NULL core will use
  * default ops
  *
  * Returns 0 if successful, or a negative error code.
  */
 int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 			const struct hdac_bus_ops *ops,
-			const struct hdac_io_ops *io_ops,
 			const struct hdac_ext_bus_ops *ext_ops)
 {
 	int ret;
 
-	/* check if io ops are provided, if not load the defaults */
-	if (io_ops == NULL)
-		io_ops = &hdac_ext_default_io;
-
-	ret = snd_hdac_bus_init(bus, dev, ops, io_ops);
+	ret = snd_hdac_bus_init(bus, dev, ops);
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 00ea12e67dc8..dc2523ef7d98 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -19,13 +19,11 @@ static const struct hdac_bus_ops default_ops = {
  * snd_hdac_bus_init - initialize a HD-audio bas bus
  * @bus: the pointer to bus object
  * @ops: bus verb operators
- * @io_ops: lowlevel I/O operators
  *
  * Returns 0 if successful, or a negative error code.
  */
 int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
-		      const struct hdac_bus_ops *ops,
-		      const struct hdac_io_ops *io_ops)
+		      const struct hdac_bus_ops *ops)
 {
 	memset(bus, 0, sizeof(*bus));
 	bus->dev = dev;
@@ -33,7 +31,6 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
 		bus->ops = ops;
 	else
 		bus->ops = &default_ops;
-	bus->io_ops = io_ops;
 	bus->dma_type = SNDRV_DMA_TYPE_DEV;
 	INIT_LIST_HEAD(&bus->stream_list);
 	INIT_LIST_HEAD(&bus->codec_list);
@@ -218,3 +215,33 @@ void snd_hdac_bus_remove_device(struct hdac_bus *bus,
 	flush_work(&bus->unsol_work);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_remove_device);
+
+#ifdef CONFIG_SND_HDA_ALIGNED_MMIO
+/* Helpers for aligned read/write of mmio space, for Tegra */
+unsigned int snd_hdac_aligned_read(void __iomem *addr, unsigned int mask)
+{
+	void __iomem *aligned_addr =
+		(void __iomem *)((unsigned long)(addr) & ~0x3);
+	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
+	unsigned int v;
+
+	v = readl(aligned_addr);
+	return (v >> shift) & mask;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_aligned_read);
+
+void snd_hdac_aligned_write(unsigned int val, void __iomem *addr,
+			    unsigned int mask)
+{
+	void __iomem *aligned_addr =
+		(void __iomem *)((unsigned long)(addr) & ~0x3);
+	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
+	unsigned int v;
+
+	v = readl(aligned_addr);
+	v &= ~(mask << shift);
+	v |= val << shift;
+	writel(v, aligned_addr);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_aligned_write);
+#endif /* CONFIG_SND_HDA_ALIGNED_MMIO */
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 35d934309cb2..82198ea8f7f8 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -26,6 +26,7 @@ config SND_HDA_TEGRA
 	tristate "NVIDIA Tegra HD Audio"
 	depends on ARCH_TEGRA
 	select SND_HDA
+	select SND_HDA_ALIGNED_MMIO
 	help
 	  Say Y here to support the HDA controller present in NVIDIA
 	  Tegra SoCs
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index c8d1b4316245..ee5504e2441f 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1202,14 +1202,12 @@ void snd_hda_bus_reset(struct hda_bus *bus)
 }
 
 /* HD-audio bus initialization */
-int azx_bus_init(struct azx *chip, const char *model,
-		 const struct hdac_io_ops *io_ops)
+int azx_bus_init(struct azx *chip, const char *model)
 {
 	struct hda_bus *bus = &chip->bus;
 	int err;
 
-	err = snd_hdac_bus_init(&bus->core, chip->card->dev, &bus_core_ops,
-				io_ops);
+	err = snd_hdac_bus_init(&bus->core, chip->card->dev, &bus_core_ops);
 	if (err < 0)
 		return err;
 
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index baa15374fbcb..146a71e0d594 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -206,8 +206,7 @@ void azx_stop_chip(struct azx *chip);
 irqreturn_t azx_interrupt(int irq, void *dev_id);
 
 /* Codec interface */
-int azx_bus_init(struct azx *chip, const char *model,
-		 const struct hdac_io_ops *io_ops);
+int azx_bus_init(struct azx *chip, const char *model);
 int azx_probe_codecs(struct azx *chip, unsigned int max_slots);
 int azx_codec_configure(struct azx *chip);
 int azx_init_streams(struct azx *chip);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3bb4c26f2799..963a92943a6d 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1627,7 +1627,6 @@ static int default_bdl_pos_adj(struct azx *chip)
 /*
  * constructor
  */
-static const struct hdac_io_ops pci_hda_io_ops;
 static const struct hda_controller_ops pci_hda_ops;
 
 static int azx_create(struct snd_card *card, struct pci_dev *pci,
@@ -1687,7 +1686,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
 	else
 		chip->bdl_pos_adj = bdl_pos_adj[dev];
 
-	err = azx_bus_init(chip, model[dev], &pci_hda_io_ops);
+	err = azx_bus_init(chip, model[dev]);
 	if (err < 0) {
 		kfree(hda);
 		pci_disable_device(pci);
@@ -1932,41 +1931,6 @@ static void azx_firmware_cb(const struct firmware *fw, void *context)
 }
 #endif
 
-/*
- * HDA controller ops.
- */
-
-/* PCI register access. */
-static void pci_azx_writel(u32 value, u32 __iomem *addr)
-{
-	writel(value, addr);
-}
-
-static u32 pci_azx_readl(u32 __iomem *addr)
-{
-	return readl(addr);
-}
-
-static void pci_azx_writew(u16 value, u16 __iomem *addr)
-{
-	writew(value, addr);
-}
-
-static u16 pci_azx_readw(u16 __iomem *addr)
-{
-	return readw(addr);
-}
-
-static void pci_azx_writeb(u8 value, u8 __iomem *addr)
-{
-	writeb(value, addr);
-}
-
-static u8 pci_azx_readb(u8 __iomem *addr)
-{
-	return readb(addr);
-}
-
 static int disable_msi_reset_irq(struct azx *chip)
 {
 	struct hdac_bus *bus = azx_bus(chip);
@@ -1994,15 +1958,6 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
 #endif
 }
 
-static const struct hdac_io_ops pci_hda_io_ops = {
-	.reg_writel = pci_azx_writel,
-	.reg_readl = pci_azx_readl,
-	.reg_writew = pci_azx_writew,
-	.reg_readw = pci_azx_readw,
-	.reg_writeb = pci_azx_writeb,
-	.reg_readb = pci_azx_readb,
-};
-
 static const struct hda_controller_ops pci_hda_ops = {
 	.disable_msi_reset_irq = disable_msi_reset_irq,
 	.pcm_mmap_prepare = pcm_mmap_prepare,
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index ba414cc639f1..8350954b7986 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -75,72 +75,6 @@ MODULE_PARM_DESC(power_save,
 #define power_save	0
 #endif
 
-/*
- * Register access ops. Tegra HDA register access is DWORD only.
- */
-static void hda_tegra_writel(u32 value, u32 __iomem *addr)
-{
-	writel(value, addr);
-}
-
-static u32 hda_tegra_readl(u32 __iomem *addr)
-{
-	return readl(addr);
-}
-
-static void hda_tegra_writew(u16 value, u16 __iomem  *addr)
-{
-	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
-	u32 v;
-
-	v = readl(dword_addr);
-	v &= ~(0xffff << shift);
-	v |= value << shift;
-	writel(v, dword_addr);
-}
-
-static u16 hda_tegra_readw(u16 __iomem *addr)
-{
-	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
-	u32 v;
-
-	v = readl(dword_addr);
-	return (v >> shift) & 0xffff;
-}
-
-static void hda_tegra_writeb(u8 value, u8 __iomem *addr)
-{
-	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
-	u32 v;
-
-	v = readl(dword_addr);
-	v &= ~(0xff << shift);
-	v |= value << shift;
-	writel(v, dword_addr);
-}
-
-static u8 hda_tegra_readb(u8 __iomem *addr)
-{
-	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
-	u32 v;
-
-	v = readl(dword_addr);
-	return (v >> shift) & 0xff;
-}
-
-static const struct hdac_io_ops hda_tegra_io_ops = {
-	.reg_writel = hda_tegra_writel,
-	.reg_readl = hda_tegra_readl,
-	.reg_writew = hda_tegra_writew,
-	.reg_readw = hda_tegra_readw,
-	.reg_writeb = hda_tegra_writeb,
-	.reg_readb = hda_tegra_readb,
-};
-
 static const struct hda_controller_ops hda_tegra_ops; /* nothing special */
 
 static void hda_tegra_init(struct hda_tegra *hda)
@@ -459,7 +393,7 @@ static int hda_tegra_create(struct snd_card *card,
 
 	INIT_WORK(&hda->probe_work, hda_tegra_probe_work);
 
-	err = azx_bus_init(chip, NULL, &hda_tegra_io_ops);
+	err = azx_bus_init(chip, NULL);
 	if (err < 0)
 		return err;
 
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 3362e71b4563..c6d8076dc2fd 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -132,7 +132,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset)
 
 	/* Reset stream-to-link mapping */
 	list_for_each_entry(hlink, &bus->hlink_list, list)
-		bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
+		writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
 
 	skl_enable_miscbdcge(bus->dev, true);
 
@@ -854,7 +854,6 @@ out_err:
  * constructor
  */
 static int skl_create(struct pci_dev *pci,
-		      const struct hdac_io_ops *io_ops,
 		      struct skl **rskl)
 {
 	struct hdac_ext_bus_ops *ext_ops = NULL;
@@ -884,7 +883,7 @@ static int skl_create(struct pci_dev *pci,
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	ext_ops = snd_soc_hdac_hda_get_ops();
 #endif
-	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);
+	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, ext_ops);
 	bus->use_posbuf = 1;
 	skl->pci = pci;
 	INIT_WORK(&skl->probe_work, skl_probe_work);
@@ -1013,7 +1012,7 @@ static int skl_probe(struct pci_dev *pci,
 	}
 
 	/* we use ext core ops, so provide NULL for ops here */
-	err = skl_create(pci, NULL, &skl);
+	err = skl_create(pci, &skl);
 	if (err < 0)
 		return err;
 
diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index 0bc93fa06b5b..438121c70f99 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -21,45 +21,6 @@ static const struct hdac_bus_ops bus_ops = {
 
 #endif
 
-static void sof_hda_writel(u32 value, u32 __iomem *addr)
-{
-	writel(value, addr);
-}
-
-static u32 sof_hda_readl(u32 __iomem *addr)
-{
-	return readl(addr);
-}
-
-static void sof_hda_writew(u16 value, u16 __iomem *addr)
-{
-	writew(value, addr);
-}
-
-static u16 sof_hda_readw(u16 __iomem *addr)
-{
-	return readw(addr);
-}
-
-static void sof_hda_writeb(u8 value, u8 __iomem *addr)
-{
-	writeb(value, addr);
-}
-
-static u8 sof_hda_readb(u8 __iomem *addr)
-{
-	return readb(addr);
-}
-
-static const struct hdac_io_ops io_ops = {
-	.reg_writel = sof_hda_writel,
-	.reg_readl = sof_hda_readl,
-	.reg_writew = sof_hda_writew,
-	.reg_readw = sof_hda_readw,
-	.reg_writeb = sof_hda_writeb,
-	.reg_readb = sof_hda_readb,
-};
-
 /*
  * This can be used for both with/without hda link support.
  */
@@ -69,7 +30,6 @@ void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev,
 	memset(bus, 0, sizeof(*bus));
 	bus->dev = dev;
 
-	bus->io_ops = &io_ops;
 	INIT_LIST_HEAD(&bus->stream_list);
 
 	bus->irq = -1;
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 91de4785b6a3..8d4ce5b4febd 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -356,7 +356,7 @@ static int hda_resume(struct snd_sof_dev *sdev)
 
 	/* Reset stream-to-link mapping */
 	list_for_each_entry(hlink, &bus->hlink_list, list)
-		bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
+		writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
 
 	hda_dsp_ctrl_misc_clock_gating(sdev, true);
 #else
-- 
cgit v1.2.3-59-g8ed1b


From d4ff1b3917a529bdc75592af6b1504ad6c4029f7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 7 Aug 2019 20:50:50 +0200
Subject: ASoC: SOF: Intel: Initialize hdaudio bus properly

The SOF HD-audio bus has its house-made initialization code.  It's
supposedly for making the code independent from HD-audio bus drivers.
However, this is error-prone, and above all, the SOF driver has
already dependency on HD-audio bus driver when CONFIG_SND_SOF_HDA is
set.  That is, if this Kconfig is set, there is no reason to avoid the
call to the proper bus init function.

Also, the ext_ops that is set at bus initialization can be better
handled inside sof_hda_bus_init().  We don't need to refer this
outside the bus initialization.

So this patch addresses these issues:
- sof_hda_bus_init() calls nothing but snd_hdac_ext_bus_init()
  when CONFIG_SND_SOF_HDA is set.  Otherwise some fields are
  initialized locally like before for avoiding the dependency.
- ext_ops is referred inside sof_hda_bus_init().  The ext_ops argument
  of snd_hda_bus_init() is dropped.

Acked-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/sof/intel/hda-bus.c | 31 +++++++++----------------------
 sound/soc/sof/intel/hda.c     |  6 +-----
 sound/soc/sof/intel/hda.h     |  3 +--
 3 files changed, 11 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index 438121c70f99..0caec3a070d3 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -12,28 +12,26 @@
 #include "../sof-priv.h"
 #include "hda.h"
 
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-
-static const struct hdac_bus_ops bus_ops = {
-	.command = snd_hdac_bus_send_cmd,
-	.get_response = snd_hdac_bus_get_response,
-};
-
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+#define sof_hda_ext_ops	snd_soc_hdac_hda_get_ops()
+#else
+#define sof_hda_ext_ops	NULL
 #endif
 
 /*
  * This can be used for both with/without hda link support.
  */
-void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev,
-		      const struct hdac_ext_bus_ops *ext_ops)
+void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev)
 {
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	snd_hdac_ext_bus_init(bus, dev, NULL, sof_hda_ext_ops);
+#else /* CONFIG_SND_SOC_SOF_HDA */
 	memset(bus, 0, sizeof(*bus));
 	bus->dev = dev;
 
 	INIT_LIST_HEAD(&bus->stream_list);
 
 	bus->irq = -1;
-	bus->ext_ops = ext_ops;
 
 	/*
 	 * There is only one HDA bus atm. keep the index as 0.
@@ -42,16 +40,5 @@ void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev,
 	bus->idx = 0;
 
 	spin_lock_init(&bus->reg_lock);
-
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-	INIT_LIST_HEAD(&bus->codec_list);
-	INIT_LIST_HEAD(&bus->hlink_list);
-
-	mutex_init(&bus->cmd_mutex);
-	mutex_init(&bus->lock);
-	bus->ops = &bus_ops;
-	INIT_WORK(&bus->unsol_work, snd_hdac_bus_process_unsol_events);
-	bus->cmd_dma_state = true;
-#endif
-
+#endif /* CONFIG_SND_SOC_SOF_HDA */
 }
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 7f665392618f..7ca27000c34d 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -236,7 +236,6 @@ static int hda_init(struct snd_sof_dev *sdev)
 {
 	struct hda_bus *hbus;
 	struct hdac_bus *bus;
-	struct hdac_ext_bus_ops *ext_ops = NULL;
 	struct pci_dev *pci = to_pci_dev(sdev->dev);
 	int ret;
 
@@ -244,10 +243,7 @@ static int hda_init(struct snd_sof_dev *sdev)
 	bus = sof_to_bus(sdev);
 
 	/* HDA bus init */
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
-	ext_ops = snd_soc_hdac_hda_get_ops();
-#endif
-	sof_hda_bus_init(bus, &pci->dev, ext_ops);
+	sof_hda_bus_init(bus, &pci->dev);
 
 	/* Workaround for a communication error on CFL (bko#199007) and CNL */
 	if (IS_CFL(pci) || IS_CNL(pci))
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index d9c17146200b..75b096050fa2 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -549,8 +549,7 @@ void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev);
 /*
  * HDA bus operations.
  */
-void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev,
-		      const struct hdac_ext_bus_ops *ext_ops);
+void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev);
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 /*
-- 
cgit v1.2.3-59-g8ed1b


From 804cbf4bb063204ca6c2471baa694548aab02ce3 Mon Sep 17 00:00:00 2001
From: Keyon Jie <yang.jie@linux.intel.com>
Date: Wed, 7 Aug 2019 09:50:30 -0500
Subject: ASoC: hdac_hda: fix page fault issue by removing race

There is a race between hda codec device removing and the
jack-detecting work, which will lead to a page fault issue as the
latter work is accessing codec device which could be already removed.

Here add the cancellation of jack-detecting work before codecs are actually
removed to avoid the race and fix the issue.

Bug: https://github.com/thesofproject/linux/issues/1067
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190807145030.26117-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdac_hda.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index 7d4940256914..91242b6f8ea7 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -495,6 +495,10 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev)
 
 static int hdac_hda_dev_remove(struct hdac_device *hdev)
 {
+	struct hdac_hda_priv *hda_pvt;
+
+	hda_pvt = dev_get_drvdata(&hdev->dev);
+	cancel_delayed_work_sync(&hda_pvt->codec.jackpoll_work);
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From e6d7020c2946bef2efab7c70339eee6a6b3cb6a6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:30:58 +0900
Subject: ASoC: soc-core: tidyup for snd_soc_add_component_controls()

snd_soc_add_component_controls() registers controls by using
for(... i < num; ...). If controls was NULL, num should be zero.
Thus, we don't need to check about controls pointer.
This patch also cares missing return value.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ef1xahor.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 10 ++++++----
 1 file changed, 6 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bb1e9e2c4ff4..e481f9999bfb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1304,10 +1304,12 @@ static int soc_probe_component(struct snd_soc_card *card,
 		}
 	}
 
-	if (component->driver->controls)
-		snd_soc_add_component_controls(component,
-					       component->driver->controls,
-					       component->driver->num_controls);
+	ret = snd_soc_add_component_controls(component,
+					     component->driver->controls,
+					     component->driver->num_controls);
+	if (ret < 0)
+		goto err_probe;
+
 	if (component->driver->dapm_routes)
 		snd_soc_dapm_add_routes(dapm,
 					component->driver->dapm_routes,
-- 
cgit v1.2.3-59-g8ed1b


From daa480bde6b3a9b728965e52efffc329ccee3f77 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:31:03 +0900
Subject: ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()

snd_soc_dapm_add_routes() registers routes by using
for(... i < num; ...). If routes was NULL, num should be zero.
Thus, we don't need to check about route pointer.
This patch also cares missing return value.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87d0hhahon.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 23 +++++++++++++----------
 1 file changed, 13 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e481f9999bfb..de95b68ce9ee 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1310,10 +1310,11 @@ static int soc_probe_component(struct snd_soc_card *card,
 	if (ret < 0)
 		goto err_probe;
 
-	if (component->driver->dapm_routes)
-		snd_soc_dapm_add_routes(dapm,
-					component->driver->dapm_routes,
-					component->driver->num_dapm_routes);
+	ret = snd_soc_dapm_add_routes(dapm,
+				      component->driver->dapm_routes,
+				      component->driver->num_dapm_routes);
+	if (ret < 0)
+		goto err_probe;
 
 	list_add(&dapm->list, &card->dapm_list);
 	/* see for_each_card_components */
@@ -2061,13 +2062,15 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		snd_soc_add_card_controls(card, card->controls,
 					  card->num_controls);
 
-	if (card->dapm_routes)
-		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
-					card->num_dapm_routes);
+	ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
+				      card->num_dapm_routes);
+	if (ret < 0)
+		goto probe_end;
 
-	if (card->of_dapm_routes)
-		snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes,
-					card->num_of_dapm_routes);
+	ret = snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes,
+				      card->num_of_dapm_routes);
+	if (ret < 0)
+		goto probe_end;
 
 	/* try to set some sane longname if DMI is available */
 	snd_soc_set_dmi_name(card, NULL);
-- 
cgit v1.2.3-59-g8ed1b


From 9b98c7c2a0599084c7ed629b5f88f474d2578307 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:31:08 +0900
Subject: ASoC: soc-core: tidyup for snd_soc_add_card_controls()

snd_soc_add_card_controls() registers controls by using
for(... i < num; ...). If controls was NULL, num should be zero.
Thus, we don't need to check about controls pointer.
This patch also cares missing return value.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87blx1ahoi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index de95b68ce9ee..4b9ae867613c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2058,9 +2058,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	snd_soc_dapm_link_dai_widgets(card);
 	snd_soc_dapm_connect_dai_link_widgets(card);
 
-	if (card->controls)
-		snd_soc_add_card_controls(card, card->controls,
-					  card->num_controls);
+	ret = snd_soc_add_card_controls(card, card->controls,
+					card->num_controls);
+	if (ret < 0)
+		goto probe_end;
 
 	ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
 				      card->num_dapm_routes);
-- 
cgit v1.2.3-59-g8ed1b


From d8ca7a0a8583fc491b625450580c4092879af3dd Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:31:14 +0900
Subject: ASoC: soc-core: call snd_soc_dapm_debugfs_init() at
 soc_init_card_debugfs()

We have 2 soc_init_card_debugfs() implementations for with/without DEBUG_FS.
But, snd_soc_instantiate_card() calls snd_soc_dapm_debugfs_init() under
ifdef DEBUG_FS after soc_init_card_debugfs(). This is very strange.
We can call snd_soc_dapm_debugfs_init() under soc_init_card_debugfs().

	#ifdef CONFIG_DEBUG_FS
=>	static void soc_init_card_debugfs(...)
	{
		...
	}
	...
	#else
=>	static inline void soc_init_card_debugfs(...)
	{
		...
	}
	#endif

	static int snd_soc_instantiate_card(struct snd_soc_card *card)
	{
		...
=>		soc_init_card_debugfs(card);

*	#ifdef CONFIG_DEBUG_FS
*		snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root);
*	#endif
	}

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a7clahob.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4b9ae867613c..cf3d967d731e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -216,6 +216,8 @@ static void soc_init_card_debugfs(struct snd_soc_card *card)
 
 	debugfs_create_u32("dapm_pop_time", 0644, card->debugfs_card_root,
 			   &card->pop_time);
+
+	snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root);
 }
 
 static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
@@ -1982,10 +1984,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 
 	soc_init_card_debugfs(card);
 
-#ifdef CONFIG_DEBUG_FS
-	snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root);
-#endif
-
 #ifdef CONFIG_PM_SLEEP
 	/* deferred resume work */
 	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
-- 
cgit v1.2.3-59-g8ed1b


From b245d273cbcd64eeaa93305f99c4ea8a6baf9c89 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:31:19 +0900
Subject: ASoC: soc-core: remove unneeded list_empty() check for
 snd_soc_try_rebind_card()

list_for_each_entry_safe() will do nothing if it was empty list.
This patch removes unneeded list_empty() check for
list_for_each_entry_safe().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878ss5aho6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index cf3d967d731e..6b0042835233 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2808,12 +2808,9 @@ static void snd_soc_try_rebind_card(void)
 {
 	struct snd_soc_card *card, *c;
 
-	if (!list_empty(&unbind_card_list)) {
-		list_for_each_entry_safe(card, c, &unbind_card_list, list) {
-			if (!snd_soc_bind_card(card))
-				list_del(&card->list);
-		}
-	}
+	list_for_each_entry_safe(card, c, &unbind_card_list, list)
+		if (!snd_soc_bind_card(card))
+			list_del(&card->list);
 }
 
 int snd_soc_add_component(struct device *dev,
-- 
cgit v1.2.3-59-g8ed1b


From b3da42519c3e6ad977af844d61c31d0e5f874f1c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:31:24 +0900
Subject: ASoC: soc-core: tidyup for card->deferred_resume_work

card->deferred_resume_work is used if CONFIG_PM_SLEEP was defined.
but
	1) It is defined even though CONFIG_PM_SLEEP was not defined
	2) random ifdef code is difficult to read.
This patch tidyup these issues.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e7paho1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  5 +++--
 sound/soc/soc-core.c | 14 ++++++++++----
 2 files changed, 13 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 6ac6481b4882..85ad971e9432 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1058,8 +1058,6 @@ struct snd_soc_card {
 	int num_of_dapm_routes;
 	bool fully_routed;
 
-	struct work_struct deferred_resume_work;
-
 	/* lists of probed devices belonging to this card */
 	struct list_head component_dev_list;
 	struct list_head list;
@@ -1079,6 +1077,9 @@ struct snd_soc_card {
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_card_root;
+#endif
+#ifdef CONFIG_PM_SLEEP
+	struct work_struct deferred_resume_work;
 #endif
 	u32 pop_time;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6b0042835233..25b26caea4e0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -701,9 +701,18 @@ int snd_soc_resume(struct device *dev)
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_resume);
+
+static void soc_resume_init(struct snd_soc_card *card)
+{
+	/* deferred resume work */
+	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
+}
 #else
 #define snd_soc_suspend NULL
 #define snd_soc_resume NULL
+static inline void soc_resume_init(struct snd_soc_card *card)
+{
+}
 #endif
 
 static const struct snd_soc_dai_ops null_dai_ops = {
@@ -1984,10 +1993,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 
 	soc_init_card_debugfs(card);
 
-#ifdef CONFIG_PM_SLEEP
-	/* deferred resume work */
-	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
-#endif
+	soc_resume_init(card);
 
 	ret = snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
 					card->num_dapm_widgets);
-- 
cgit v1.2.3-59-g8ed1b


From ee5b3f11416d1ba69e919b2fe86aae0b46f9a83e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:31:31 +0900
Subject: ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS

soc_dpcm_debugfs_add() is implemented at soc-pcm.c under CONFIG_DEBUG_FS.
Thus, soc-core.c which is only user of it need to use CONFIG_DEBUG_FS, too.

This patch defines soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS case.
Then, we can remove #ifdef CONFIG_DEBUG_FS from soc-core.c

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875zn9ahnv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dpcm.h | 9 ++++++++-
 sound/soc/soc-core.c     | 2 --
 2 files changed, 8 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index 4be3a2b7c106..e55aeb00ce2d 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -142,9 +142,16 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
 
 /* internal use only */
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
-void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
 int soc_dpcm_runtime_update(struct snd_soc_card *);
 
+#ifdef CONFIG_DEBUG_FS
+void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
+#else
+static inline void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
+{
+}
+#endif
+
 int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
 	int stream, struct snd_soc_dapm_widget_list **list_);
 int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 25b26caea4e0..2a75fba31aa4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1487,11 +1487,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 	if (ret)
 		return ret;
 
-#ifdef CONFIG_DEBUG_FS
 	/* add DPCM sysfs entries */
 	if (dai_link->dynamic)
 		soc_dpcm_debugfs_add(rtd);
-#endif
 
 	num = rtd->num;
 
-- 
cgit v1.2.3-59-g8ed1b


From 596becd3f82a7b7e091b0f5c380bc9a0e6758126 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 7 Aug 2019 10:31:36 +0900
Subject: ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add()

soc_dpcm_debugfs_add(rtd) is checking rtd->dai_link pointer,
but, rtd->dai_link->dynamic have been already checked before calling it.

	static int soc_probe_link_dais(...) {
		dai_link = rtd->dai_link;
		...
=>		if (dai_link->dynamic)
=>			soc_dpcm_debugfs_add(rtd);
		...
	}

	void soc_dpcm_debugfs_add(rtd)
	{
=>		if (!rtd->dai_link)
			return;
		...
	}

These pointer checks are strange/pointless.
This patch checks dai_link->dynamic under soc_dpcm_debugfs_add().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874l2tahnq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 3 +--
 sound/soc/soc-pcm.c  | 3 +++
 2 files changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2a75fba31aa4..1fbd525763d5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1488,8 +1488,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		return ret;
 
 	/* add DPCM sysfs entries */
-	if (dai_link->dynamic)
-		soc_dpcm_debugfs_add(rtd);
+	soc_dpcm_debugfs_add(rtd);
 
 	num = rtd->num;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 77c986fe08d0..da657c8179cc 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -3200,6 +3200,9 @@ void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
 	if (!rtd->dai_link)
 		return;
 
+	if (!rtd->dai_link->dynamic)
+		return;
+
 	if (!rtd->card->debugfs_card_root)
 		return;
 
-- 
cgit v1.2.3-59-g8ed1b


From 0e36f36b04e7677c6b74349c9d4baea1ed5668f2 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 7 Aug 2019 21:51:31 -0500
Subject: ASoC: soc-core: fix module_put() warning in soc_cleanup_component

The recent changes introduce warnings in the SOF load/unload module
tests. The code does not seem balanced with a confusion between
_close() and _remove() macros. Using _remove() fixes the issue and
removes the warning.

Suggested-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Fixes: 4a81e8f30d0b4 ('ASoC: soc-component: add snd_soc_component_get/put()')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20190808025131.32482-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1fbd525763d5..bf45e60eb34f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -935,7 +935,7 @@ static void soc_cleanup_component(struct snd_soc_component *component)
 	snd_soc_dapm_free(snd_soc_component_get_dapm(component));
 	soc_cleanup_component_debugfs(component);
 	component->card = NULL;
-	snd_soc_component_module_put_when_close(component);
+	snd_soc_component_module_put_when_remove(component);
 }
 
 static void soc_remove_component(struct snd_soc_component *component)
-- 
cgit v1.2.3-59-g8ed1b


From 8e082d8f42fabf9a4a0708d8012f4995765478fc Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 8 Aug 2019 11:25:52 +0800
Subject: ASoC: tscs454: remove unused variable 'PLL_48K_RATE'

The global variable 'PLL_48K_RATE' is never used
so just remove it.

Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190808032552.45360-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tscs454.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c
index 93d84e5ae2d5..c3587af9985c 100644
--- a/sound/soc/codecs/tscs454.c
+++ b/sound/soc/codecs/tscs454.c
@@ -22,7 +22,6 @@
 
 #include "tscs454.h"
 
-static const unsigned int PLL_48K_RATE = (48000 * 256);
 static const unsigned int PLL_44_1K_RATE = (44100 * 256);
 
 #define COEFF_SIZE 3
-- 
cgit v1.2.3-59-g8ed1b


From 0e4cc44f5066b22938f384cd57db99d66311de90 Mon Sep 17 00:00:00 2001
From: Libin Yang <libin.yang@intel.com>
Date: Thu, 8 Aug 2019 14:27:34 -0500
Subject: ASoC: intel: skl_hda_dsp_common: create HDMI jack kctl

This patch call snd_jack_add_new_kctl() to create the HDMI jack kctls.
Userspace needs these kctls to detect the hdmi monitor hotplug.

In /usr/share/alsa/ucm, the config file needs to assign a jack kctl to
"JackControl" to let PA get the jack hotplug status.

Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190808192734.18286-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/skl_hda_dsp_common.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c
index 55fd82e05e2c..58409b6e476e 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_common.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.c
@@ -147,6 +147,11 @@ int skl_hda_hdmi_jack_init(struct snd_soc_card *card)
 		if (err)
 			return err;
 
+		err = snd_jack_add_new_kctl(pcm->hdmi_jack.jack,
+					    jack_name, SND_JACK_AVOUT);
+		if (err)
+			dev_warn(component->dev, "failed creating Jack kctl\n");
+
 		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
 					  &pcm->hdmi_jack);
 		if (err < 0)
-- 
cgit v1.2.3-59-g8ed1b


From 63e45324c284d240562a5b435afc4b23ae7d923c Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 8 Aug 2019 20:15:48 +0200
Subject: ASoC: Intel: Skylake: Limit large_config_get to single frame

Reply for the very first LARGE_CONFIG_GET request contains total size of
payload to be retrieved by host.
From then on, each subsequent reply carries buffer offset instead. As
looping is not covered by any real-life example, remove it and cleanup
the function for followup overhaul.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190808181549.12521-2-cezary.rojewski@intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-sst-ipc.c | 37 +++++++----------------------------
 1 file changed, 7 insertions(+), 30 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index a2b69a02aab2..196c80dadb1f 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -973,8 +973,7 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
 {
 	struct skl_ipc_header header = {0};
 	struct sst_ipc_message request = {0}, reply = {0};
-	int ret = 0;
-	size_t sz_remaining, rx_size, data_offset;
+	int ret;
 
 	header.primary = IPC_MSG_TARGET(IPC_MOD_MSG);
 	header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST);
@@ -987,34 +986,12 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
 	header.extension |= IPC_FINAL_BLOCK(1);
 	header.extension |= IPC_INITIAL_BLOCK(1);
 
-	sz_remaining = msg->param_data_size;
-	data_offset = 0;
-
-	while (sz_remaining != 0) {
-		rx_size = sz_remaining > SKL_ADSP_W1_SZ
-				? SKL_ADSP_W1_SZ : sz_remaining;
-		if (rx_size == sz_remaining)
-			header.extension |= IPC_FINAL_BLOCK(1);
-
-		request.header = *(u64 *)(&header);
-		reply.data = ((char *)param) + data_offset;
-		reply.size = msg->param_data_size;
-		ret = sst_ipc_tx_message_wait(ipc, request, &reply);
-		if (ret < 0) {
-			dev_err(ipc->dev,
-				"ipc: get large config fail, err: %d\n", ret);
-			return ret;
-		}
-		sz_remaining -= rx_size;
-		data_offset = msg->param_data_size - sz_remaining;
-
-		/* clear the fields */
-		header.extension &= IPC_INITIAL_BLOCK_CLEAR;
-		header.extension &= IPC_DATA_OFFSET_SZ_CLEAR;
-		/* fill the fields */
-		header.extension |= IPC_INITIAL_BLOCK(1);
-		header.extension |= IPC_DATA_OFFSET_SZ(data_offset);
-	}
+	request.header = *(u64 *)(&header);
+	reply.data = param;
+	reply.size = msg->param_data_size;
+	ret = sst_ipc_tx_message_wait(ipc, request, &reply);
+	if (ret < 0)
+		dev_err(ipc->dev, "ipc: get large config fail, err: %d\n", ret);
 
 	return ret;
 }
-- 
cgit v1.2.3-59-g8ed1b


From 36004c42c761a6d95b867a4fbb9a955034e53351 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 8 Aug 2019 20:15:49 +0200
Subject: ASoC: Intel: Skylake: large_config_get overhaul

LARGE_CONFIG_GET is mainly used to retrieve requested module parameters
but it may also carry TX payload with them. Update its implementation to
account for both TX and RX data.
First reply.header carries total payload size within data_off_sizefield.
Make use of reply.header to realloc returned buffer with correct size.

Failure of IPC request is permissive - error-payload may be returned, an
informative data why GET for given param failed - and thus function
should not collapse before entire processing is finished. Caller is
responsible for checking returned payload and bytes parameters.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190808181549.12521-3-cezary.rojewski@intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c |  3 ++-
 sound/soc/intel/skylake/skl-sst-ipc.c  | 25 ++++++++++++++++++++-----
 sound/soc/intel/skylake/skl-sst-ipc.h  |  3 ++-
 3 files changed, 24 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index e8cc710f092b..84f0e6f58eb5 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -1379,11 +1379,12 @@ int skl_get_module_params(struct skl_dev *skl, u32 *params, int size,
 			  u32 param_id, struct skl_module_cfg *mcfg)
 {
 	struct skl_ipc_large_config_msg msg;
+	size_t bytes = size;
 
 	msg.module_id = mcfg->id.module_id;
 	msg.instance_id = mcfg->id.pvt_id;
 	msg.param_data_size = size;
 	msg.large_param_id = param_id;
 
-	return skl_ipc_get_large_config(&skl->ipc, &msg, params);
+	return skl_ipc_get_large_config(&skl->ipc, &msg, &params, &bytes);
 }
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index 196c80dadb1f..667cdddc289f 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -969,12 +969,18 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc,
 EXPORT_SYMBOL_GPL(skl_ipc_set_large_config);
 
 int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
-		struct skl_ipc_large_config_msg *msg, u32 *param)
+		struct skl_ipc_large_config_msg *msg,
+		u32 **payload, size_t *bytes)
 {
 	struct skl_ipc_header header = {0};
-	struct sst_ipc_message request = {0}, reply = {0};
+	struct sst_ipc_message request, reply = {0};
+	unsigned int *buf;
 	int ret;
 
+	reply.data = kzalloc(SKL_ADSP_W1_SZ, GFP_KERNEL);
+	if (!reply.data)
+		return -ENOMEM;
+
 	header.primary = IPC_MSG_TARGET(IPC_MOD_MSG);
 	header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST);
 	header.primary |= IPC_GLB_TYPE(IPC_MOD_LARGE_CONFIG_GET);
@@ -986,13 +992,22 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
 	header.extension |= IPC_FINAL_BLOCK(1);
 	header.extension |= IPC_INITIAL_BLOCK(1);
 
-	request.header = *(u64 *)(&header);
-	reply.data = param;
-	reply.size = msg->param_data_size;
+	request.header = *(u64 *)&header;
+	request.data = *payload;
+	request.size = *bytes;
+	reply.size = SKL_ADSP_W1_SZ;
+
 	ret = sst_ipc_tx_message_wait(ipc, request, &reply);
 	if (ret < 0)
 		dev_err(ipc->dev, "ipc: get large config fail, err: %d\n", ret);
 
+	reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK;
+	buf = krealloc(reply.data, reply.size, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+	*payload = buf;
+	*bytes = reply.size;
+
 	return ret;
 }
 EXPORT_SYMBOL_GPL(skl_ipc_get_large_config);
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h
index 93af08cf41d2..08ac31778325 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.h
+++ b/sound/soc/intel/skylake/skl-sst-ipc.h
@@ -139,7 +139,8 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc,
 		struct skl_ipc_large_config_msg *msg, u32 *param);
 
 int skl_ipc_get_large_config(struct sst_generic_ipc *ipc,
-		struct skl_ipc_large_config_msg *msg, u32 *param);
+		struct skl_ipc_large_config_msg *msg,
+		u32 **payload, size_t *bytes);
 
 int skl_sst_ipc_load_library(struct sst_generic_ipc *ipc,
 			u8 dma_id, u8 table_id, bool wait);
-- 
cgit v1.2.3-59-g8ed1b


From f6326fa4855b0f8173af822fcc139afdbf4d4b71 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Thu, 8 Aug 2019 23:15:54 +0100
Subject: ASoC: sof: Fix warning when IPC flood test is not enabled

dentry is only used when the flood test is done so move the declaration
of the variable inside the ifdef for the flood test.

Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/debug.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index 40940b2fe9d5..54cd431faab7 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -221,7 +221,6 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer,
 {
 	struct snd_sof_dfsentry *dfse = file->private_data;
 	struct snd_sof_dev *sdev = dfse->sdev;
-	struct dentry *dentry;
 	loff_t pos = *ppos;
 	size_t size_ret;
 	int skip = 0;
@@ -229,6 +228,8 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer,
 	u8 *buf;
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
+	struct dentry *dentry;
+
 	dentry = file->f_path.dentry;
 	if ((!strcmp(dentry->d_name.name, "ipc_flood_count") ||
 	     !strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) &&
-- 
cgit v1.2.3-59-g8ed1b


From fe4010667741df8c8d6bef1780ec34ebf43cf007 Mon Sep 17 00:00:00 2001
From: Stephen Rothwell <sfr@canb.auug.org.au>
Date: Fri, 9 Aug 2019 12:54:58 +1000
Subject: ALSA: hda: readl/writel need linux/io.h

Fixes: 19abfefd4c76 ("ALSA: hda: Direct MMIO accesses")
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_bus.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index dc2523ef7d98..cd25e2b3f7f2 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -4,6 +4,7 @@
  */
 
 #include <linux/init.h>
+#include <linux/io.h>
 #include <linux/device.h>
 #include <linux/module.h>
 #include <linux/export.h>
-- 
cgit v1.2.3-59-g8ed1b


From cc911c2251f07470f8e085ec8ff670e6bdd56add Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Fri, 9 Aug 2019 17:01:13 +0800
Subject: ALSA: ac97: remove unused variable 'snd_ac97_controls_master_mono'

sound/pci/ac97/ac97_codec.c:599:38: warning:
 snd_ac97_controls_master_mono defined but not used [-Wunused-const-variable=]

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_codec.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 96b4601aae73..66f6c3bf08e3 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -596,11 +596,6 @@ static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol,
 	return err;
 }
 
-static const struct snd_kcontrol_new snd_ac97_controls_master_mono[2] = {
-AC97_SINGLE("Master Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
-AC97_SINGLE("Master Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1)
-};
-
 static const struct snd_kcontrol_new snd_ac97_controls_tone[2] = {
 AC97_SINGLE("Tone Control - Bass", AC97_MASTER_TONE, 8, 15, 1),
 AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1)
-- 
cgit v1.2.3-59-g8ed1b


From a62bd63893027bfa32fccbba0e0ac067824c362c Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Fri, 9 Aug 2019 19:01:00 +0800
Subject: ASoC: SOF: Intel: Add missing include file hdac_hda.h

Building with SND_SOC_SOF_HDA_AUDIO_CODEC fails:

sound/soc/sof/intel/hda-bus.c: In function sof_hda_bus_init:
sound/soc/sof/intel/hda-bus.c:16:25: error: implicit declaration of function
 snd_soc_hdac_hda_get_ops; did you mean snd_soc_jack_add_gpiods? [-Werror=implicit-function-declaration]
 #define sof_hda_ext_ops snd_soc_hdac_hda_get_ops()

Reported-by: Hulk Robot <hulkci@huawei.com>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Fixes: d4ff1b3917a5 ('ASoC: SOF: Intel: Initialize hdaudio bus properly")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190809110100.71236-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-bus.c | 1 +
 sound/soc/sof/intel/hda.c     | 3 ---
 2 files changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index 0caec3a070d3..1d2babdda9dd 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -13,6 +13,7 @@
 #include "hda.h"
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+#include "../../codecs/hdac_hda.h"
 #define sof_hda_ext_ops	snd_soc_hdac_hda_get_ops()
 #else
 #define sof_hda_ext_ops	NULL
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index d04844d6b104..28eb780494aa 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -23,9 +23,6 @@
 #include <sound/sof/xtensa.h>
 #include "../ops.h"
 #include "hda.h"
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
-#include "../../codecs/hdac_hda.h"
-#endif
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 #include <sound/soc-acpi-intel-match.h>
-- 
cgit v1.2.3-59-g8ed1b


From bc8d9f737fc01cce913f1cc215b7e66f01697e52 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Fri, 9 Aug 2019 16:02:34 +0800
Subject: ASoC: mt6351: remove unused variable 'mt_lineout_control'

sound/soc/codecs/mt6351.c:1070:38: warning:
 mt_lineout_control defined but not used [-Wunused-const-variable=]

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190809080234.23332-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/mt6351.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/mt6351.c b/sound/soc/codecs/mt6351.c
index 4b3ce01c5a93..5c0536eb1044 100644
--- a/sound/soc/codecs/mt6351.c
+++ b/sound/soc/codecs/mt6351.c
@@ -1066,11 +1066,6 @@ static int mt_mic_bias_2_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-/* DAPM Kcontrols */
-static const struct snd_kcontrol_new mt_lineout_control =
-	SOC_DAPM_SINGLE("Switch", MT6351_AUDDEC_ANA_CON3,
-			RG_AUDLOLPWRUP_VAUDP32_BIT, 1, 0);
-
 /* DAPM Widgets */
 static const struct snd_soc_dapm_widget mt6351_dapm_widgets[] = {
 	/* Digital Clock */
-- 
cgit v1.2.3-59-g8ed1b


From 0fd70e22a0ffebd13028bf2c7da6b747070475bf Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Fri, 9 Aug 2019 16:24:40 +0800
Subject: ASoC: ml26124: remove unused variable 'ngth'

In file included from ./include/sound/tlv.h:10:0,
                 from sound/soc/codecs/ml26124.c:19:
sound/soc/codecs/ml26124.c:59:35: warning: ngth defined but not used [-Wunused-const-variable=]
 static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
                                   ^
./include/uapi/sound/tlv.h:64:15: note: in definition of macro SNDRV_CTL_TLVD_DECLARE_DB_SCALE
  unsigned int name[] = { \
               ^~~~
sound/soc/codecs/ml26124.c:59:14: note: in expansion of macro DECLARE_TLV_DB_SCALE
 static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
              ^~~~~~~~~~~~~~~~~~~~

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190809082440.67412-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ml26124.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 3abd27893ce6..55823bc95d06 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -56,7 +56,6 @@ static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0);
 static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0);
 static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0);
 static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0);
-static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
 
 static const char * const ml26124_companding[] = {"16bit PCM", "u-law",
 						  "A-law"};
-- 
cgit v1.2.3-59-g8ed1b


From 0b76f512c206dd7c3f7761b24826887af80204c6 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 7 Aug 2019 10:01:58 -0500
Subject: ASoC: SOF: Remove call to snd_sof_dsp_mailbox_init

This is reserved for some historical reason, we didn't enable memory
windows for byt/bdw at the beginning, to make it compatible, we get
those mailbox offsets from fw_ready struct firstly, and then update them
if they existed in the following memory windows, to make sure the
mailbox still can be used if no memory windows are created.

With this change all platforms have the same implementation for
xxx_fw_ready function so that we can refactor it in a common file.

Suggested-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190807150203.26359-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/bdw.c | 5 -----
 sound/soc/sof/intel/byt.c | 5 -----
 2 files changed, 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 4bb9636da990..94be024a99eb 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -455,11 +455,6 @@ static int bdw_fw_ready(struct snd_sof_dev *sdev, u32 msg_id)
 	sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready,
 		       sizeof(*fw_ready));
 
-	snd_sof_dsp_mailbox_init(sdev, fw_ready->dspbox_offset,
-				 fw_ready->dspbox_size,
-				 fw_ready->hostbox_offset,
-				 fw_ready->hostbox_size);
-
 	/* make sure ABI version is compatible */
 	ret = snd_sof_ipc_valid(sdev);
 	if (ret < 0)
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 000d576f6a8d..cb8f02ee09f2 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -237,11 +237,6 @@ static int byt_fw_ready(struct snd_sof_dev *sdev, u32 msg_id)
 	sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready,
 		       sizeof(*fw_ready));
 
-	snd_sof_dsp_mailbox_init(sdev, fw_ready->dspbox_offset,
-				 fw_ready->dspbox_size,
-				 fw_ready->hostbox_offset,
-				 fw_ready->hostbox_size);
-
 	/* make sure ABI version is compatible */
 	ret = snd_sof_ipc_valid(sdev);
 	if (ret < 0)
-- 
cgit v1.2.3-59-g8ed1b


From ddf14b640b05839cfe44d686d5a5cc7c55076094 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 7 Aug 2019 10:02:02 -0500
Subject: ASoC: SOF: Intel: bdw: Use generic function for fw ready / mem
 windows creation

bdw_get_windows / bdw_fw_ready is identical with the generic
implementation introduced in a previous patch.

So remove bdw_get_windows / bdw_fw_ready and use the generic
sof_get_windows version.

Do not forget to implement get_mailbox_offset/get_window_offset
so that we export the correct mailbox/memory window offset to
the outside world.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190807150203.26359-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/bdw.c | 156 ++++------------------------------------------
 1 file changed, 13 insertions(+), 143 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 94be024a99eb..e282179263e8 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -327,148 +327,6 @@ static irqreturn_t bdw_irq_thread(int irq, void *context)
 	return IRQ_HANDLED;
 }
 
-/*
- * IPC Firmware ready.
- */
-static void bdw_get_windows(struct snd_sof_dev *sdev)
-{
-	struct sof_ipc_window_elem *elem;
-	u32 outbox_offset = 0;
-	u32 stream_offset = 0;
-	u32 inbox_offset = 0;
-	u32 outbox_size = 0;
-	u32 stream_size = 0;
-	u32 inbox_size = 0;
-	int i;
-
-	if (!sdev->info_window) {
-		dev_err(sdev->dev, "error: have no window info\n");
-		return;
-	}
-
-	for (i = 0; i < sdev->info_window->num_windows; i++) {
-		elem = &sdev->info_window->window[i];
-
-		switch (elem->type) {
-		case SOF_IPC_REGION_UPBOX:
-			inbox_offset = elem->offset + MBOX_OFFSET;
-			inbox_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BDW_DSP_BAR] +
-						inbox_offset,
-						elem->size, "inbox",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_DOWNBOX:
-			outbox_offset = elem->offset + MBOX_OFFSET;
-			outbox_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BDW_DSP_BAR] +
-						outbox_offset,
-						elem->size, "outbox",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_TRACE:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BDW_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "etrace",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_DEBUG:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BDW_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "debug",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_STREAM:
-			stream_offset = elem->offset + MBOX_OFFSET;
-			stream_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BDW_DSP_BAR] +
-						stream_offset,
-						elem->size, "stream",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_REGS:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BDW_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "regs",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_EXCEPTION:
-			sdev->dsp_oops_offset = elem->offset + MBOX_OFFSET;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BDW_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "exception",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		default:
-			dev_err(sdev->dev, "error: get illegal window info\n");
-			return;
-		}
-	}
-
-	if (outbox_size == 0 || inbox_size == 0) {
-		dev_err(sdev->dev, "error: get illegal mailbox window\n");
-		return;
-	}
-
-	snd_sof_dsp_mailbox_init(sdev, inbox_offset, inbox_size,
-				 outbox_offset, outbox_size);
-	sdev->stream_box.offset = stream_offset;
-	sdev->stream_box.size = stream_size;
-
-	dev_dbg(sdev->dev, " mailbox upstream 0x%x - size 0x%x\n",
-		inbox_offset, inbox_size);
-	dev_dbg(sdev->dev, " mailbox downstream 0x%x - size 0x%x\n",
-		outbox_offset, outbox_size);
-	dev_dbg(sdev->dev, " stream region 0x%x - size 0x%x\n",
-		stream_offset, stream_size);
-}
-
-/* check for ABI compatibility and create memory windows on first boot */
-static int bdw_fw_ready(struct snd_sof_dev *sdev, u32 msg_id)
-{
-	struct sof_ipc_fw_ready *fw_ready = &sdev->fw_ready;
-	u32 offset;
-	int ret;
-
-	/* mailbox must be on 4k boundary */
-	offset = MBOX_OFFSET;
-
-	dev_dbg(sdev->dev, "ipc: DSP is ready 0x%8.8x offset %d\n",
-		msg_id, offset);
-
-	/* no need to re-check version/ABI for subsequent boots */
-	if (!sdev->first_boot)
-		return 0;
-
-	/* copy data from the DSP FW ready offset */
-	sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready,
-		       sizeof(*fw_ready));
-
-	/* make sure ABI version is compatible */
-	ret = snd_sof_ipc_valid(sdev);
-	if (ret < 0)
-		return ret;
-
-	/* now check for extended data */
-	snd_sof_fw_parse_ext_data(sdev, sdev->mmio_bar, MBOX_OFFSET +
-				  sizeof(struct sof_ipc_fw_ready));
-
-	bdw_get_windows(sdev);
-
-	return 0;
-}
-
 /*
  * IPC Mailbox IO
  */
@@ -522,6 +380,16 @@ static void bdw_get_reply(struct snd_sof_dev *sdev)
 	msg->reply_error = ret;
 }
 
+static int bdw_get_mailbox_offset(struct snd_sof_dev *sdev)
+{
+	return MBOX_OFFSET;
+}
+
+static int bdw_get_window_offset(struct snd_sof_dev *sdev, u32 id)
+{
+	return MBOX_OFFSET;
+}
+
 static void bdw_host_done(struct snd_sof_dev *sdev)
 {
 	/* clear BUSY bit and set DONE bit - accept new messages */
@@ -672,7 +540,9 @@ const struct snd_sof_dsp_ops sof_bdw_ops = {
 
 	/* ipc */
 	.send_msg	= bdw_send_msg,
-	.fw_ready	= bdw_fw_ready,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset = bdw_get_mailbox_offset,
+	.get_window_offset = bdw_get_window_offset,
 
 	.ipc_msg_data	= intel_ipc_msg_data,
 	.ipc_pcm_params	= intel_ipc_pcm_params,
-- 
cgit v1.2.3-59-g8ed1b


From 6eebd3907527764618ee250e1106b9e809c46143 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 7 Aug 2019 10:02:03 -0500
Subject: ASoC: SOF: Intel: hda: Use generic function for fw ready / mem
 windows creation

We can use generic sof_fw_ready function and reduce code duplication.
Careful here that we need to provide the implementation for
get_mailbox_offset and get_window_offset.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190807150203.26359-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/apl.c     |   4 +-
 sound/soc/sof/intel/cnl.c     |   4 +-
 sound/soc/sof/intel/hda-ipc.c | 150 ++----------------------------------------
 sound/soc/sof/intel/hda.h     |   4 +-
 4 files changed, 13 insertions(+), 149 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c
index fd2e26d79796..8dc7a5558da4 100644
--- a/sound/soc/sof/intel/apl.c
+++ b/sound/soc/sof/intel/apl.c
@@ -46,7 +46,9 @@ const struct snd_sof_dsp_ops sof_apl_ops = {
 
 	/* ipc */
 	.send_msg	= hda_dsp_ipc_send_msg,
-	.fw_ready	= hda_dsp_ipc_fw_ready,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset = hda_dsp_ipc_get_mailbox_offset,
+	.get_window_offset = hda_dsp_ipc_get_window_offset,
 
 	.ipc_msg_data	= hda_ipc_msg_data,
 	.ipc_pcm_params	= hda_ipc_pcm_params,
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index ffd8d4394537..6d7d9c93252c 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -204,7 +204,9 @@ const struct snd_sof_dsp_ops sof_cnl_ops = {
 
 	/* ipc */
 	.send_msg	= cnl_ipc_send_msg,
-	.fw_ready	= hda_dsp_ipc_fw_ready,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset = hda_dsp_ipc_get_mailbox_offset,
+	.get_window_offset = hda_dsp_ipc_get_window_offset,
 
 	.ipc_msg_data	= hda_ipc_msg_data,
 	.ipc_pcm_params	= hda_ipc_pcm_params,
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 2ecba91f5219..6aae6f18b3dc 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -266,156 +266,14 @@ out:
 	return ret;
 }
 
-/* IPC Firmware ready */
-
-static void ipc_get_windows(struct snd_sof_dev *sdev)
+int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev)
 {
-	struct sof_ipc_window_elem *elem;
-	u32 outbox_offset = 0;
-	u32 stream_offset = 0;
-	u32 inbox_offset = 0;
-	u32 outbox_size = 0;
-	u32 stream_size = 0;
-	u32 inbox_size = 0;
-	int i;
-
-	if (!sdev->info_window) {
-		dev_err(sdev->dev, "error: have no window info\n");
-		return;
-	}
-
-	for (i = 0; i < sdev->info_window->num_windows; i++) {
-		elem = &sdev->info_window->window[i];
-
-		switch (elem->type) {
-		case SOF_IPC_REGION_UPBOX:
-			inbox_offset =
-				elem->offset + SRAM_WINDOW_OFFSET(elem->id);
-			inbox_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[HDA_DSP_BAR] +
-						inbox_offset,
-						elem->size, "inbox",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_DOWNBOX:
-			outbox_offset =
-				elem->offset + SRAM_WINDOW_OFFSET(elem->id);
-			outbox_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[HDA_DSP_BAR] +
-						outbox_offset,
-						elem->size, "outbox",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_TRACE:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[HDA_DSP_BAR] +
-						elem->offset +
-						SRAM_WINDOW_OFFSET
-						(elem->id),
-						elem->size, "etrace",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_DEBUG:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[HDA_DSP_BAR] +
-						elem->offset +
-						SRAM_WINDOW_OFFSET
-						(elem->id),
-						elem->size, "debug",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_STREAM:
-			stream_offset =
-				elem->offset + SRAM_WINDOW_OFFSET(elem->id);
-			stream_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[HDA_DSP_BAR] +
-						elem->offset +
-						SRAM_WINDOW_OFFSET
-						(elem->id),
-						elem->size, "stream",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_REGS:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[HDA_DSP_BAR] +
-						elem->offset +
-						SRAM_WINDOW_OFFSET
-						(elem->id),
-						elem->size, "regs",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_EXCEPTION:
-			sdev->dsp_oops_offset = elem->offset +
-						SRAM_WINDOW_OFFSET(elem->id);
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[HDA_DSP_BAR] +
-						elem->offset +
-						SRAM_WINDOW_OFFSET
-						(elem->id),
-						elem->size, "exception",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		default:
-			dev_err(sdev->dev, "error: get illegal window info\n");
-			return;
-		}
-	}
-
-	if (outbox_size == 0 || inbox_size == 0) {
-		dev_err(sdev->dev, "error: get illegal mailbox window\n");
-		return;
-	}
-
-	snd_sof_dsp_mailbox_init(sdev, inbox_offset, inbox_size,
-				 outbox_offset, outbox_size);
-	sdev->stream_box.offset = stream_offset;
-	sdev->stream_box.size = stream_size;
-
-	dev_dbg(sdev->dev, " mailbox upstream 0x%x - size 0x%x\n",
-		inbox_offset, inbox_size);
-	dev_dbg(sdev->dev, " mailbox downstream 0x%x - size 0x%x\n",
-		outbox_offset, outbox_size);
-	dev_dbg(sdev->dev, " stream region 0x%x - size 0x%x\n",
-		stream_offset, stream_size);
+	return HDA_DSP_MBOX_UPLINK_OFFSET;
 }
 
-/* check for ABI compatibility and create memory windows on first boot */
-int hda_dsp_ipc_fw_ready(struct snd_sof_dev *sdev, u32 msg_id)
+int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id)
 {
-	struct sof_ipc_fw_ready *fw_ready = &sdev->fw_ready;
-	u32 offset;
-	int ret;
-
-	/* mailbox must be on 4k boundary */
-	offset = HDA_DSP_MBOX_UPLINK_OFFSET;
-
-	dev_dbg(sdev->dev, "ipc: DSP is ready 0x%8.8x offset 0x%x\n",
-		msg_id, offset);
-
-	/* no need to re-check version/ABI for subsequent boots */
-	if (!sdev->first_boot)
-		return 0;
-
-	/* copy data from the DSP FW ready offset */
-	sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready,
-		       sizeof(*fw_ready));
-
-	/* make sure ABI version is compatible */
-	ret = snd_sof_ipc_valid(sdev);
-	if (ret < 0)
-		return ret;
-
-	/* now check for extended data */
-	snd_sof_fw_parse_ext_data(sdev, sdev->mmio_bar,
-				  HDA_DSP_MBOX_UPLINK_OFFSET +
-				  sizeof(struct sof_ipc_fw_ready));
-
-	ipc_get_windows(sdev);
-
-	return 0;
+	return SRAM_WINDOW_OFFSET(id);
 }
 
 void hda_ipc_msg_data(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 65904c3511df..6db3dcbab3bd 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -519,7 +519,9 @@ int hda_ipc_pcm_params(struct snd_sof_dev *sdev,
 int hda_dsp_ipc_send_msg(struct snd_sof_dev *sdev,
 			 struct snd_sof_ipc_msg *msg);
 void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev);
-int hda_dsp_ipc_fw_ready(struct snd_sof_dev *sdev, u32 msg_id);
+int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev);
+int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id);
+
 irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context);
 irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context);
 int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir);
-- 
cgit v1.2.3-59-g8ed1b


From bb9c93f58cb66049cb5a5570cc476d9a8a97f491 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 7 Aug 2019 10:01:59 -0500
Subject: ASoC: SOF: Introduce snd_sof_dsp_get_mailbox_offset

This will allow us to export mailbox offset in order to
read the fw_ready message from.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190807150203.26359-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/ops.h      | 9 +++++++++
 sound/soc/sof/sof-priv.h | 2 ++
 2 files changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index 793c1aea0c53..c820606f19a1 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -119,6 +119,15 @@ static inline int snd_sof_dsp_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 	return sdev->mmio_bar;
 }
 
+static inline int snd_sof_dsp_get_mailbox_offset(struct snd_sof_dev *sdev)
+{
+	if (sof_ops(sdev)->get_mailbox_offset)
+		return sof_ops(sdev)->get_mailbox_offset(sdev);
+
+	dev_err(sdev->dev, "error: %s not defined\n", __func__);
+	return -ENOTSUPP;
+}
+
 /* power management */
 static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev)
 {
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index 1cec3f23f9cd..3aefb8ec3ef9 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -198,6 +198,8 @@ struct snd_sof_dsp_ops {
 	/* misc */
 	int (*get_bar_index)(struct snd_sof_dev *sdev,
 			     u32 type); /* optional */
+	int (*get_mailbox_offset)(struct snd_sof_dev *sdev);/* mandatory for common loader code */
+
 	/* DAI ops */
 	struct snd_soc_dai_driver *drv;
 	int num_drv;
-- 
cgit v1.2.3-59-g8ed1b


From e17422cda9d60339d71e117d3fdfd444e83669a4 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 7 Aug 2019 10:02:00 -0500
Subject: ASoC: SOF: Introduce snd_sof_dsp_get_window_offset

This will allow us to export the offset for a memory window.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190807150203.26359-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/ops.h      | 9 +++++++++
 sound/soc/sof/sof-priv.h | 2 ++
 2 files changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index c820606f19a1..824d36fe59fd 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -128,6 +128,15 @@ static inline int snd_sof_dsp_get_mailbox_offset(struct snd_sof_dev *sdev)
 	return -ENOTSUPP;
 }
 
+static inline int snd_sof_dsp_get_window_offset(struct snd_sof_dev *sdev,
+						u32 id)
+{
+	if (sof_ops(sdev)->get_window_offset)
+		return sof_ops(sdev)->get_window_offset(sdev, id);
+
+	dev_err(sdev->dev, "error: %s not defined\n", __func__);
+	return -ENOTSUPP;
+}
 /* power management */
 static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev)
 {
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index 3aefb8ec3ef9..de11f28bae3a 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -199,6 +199,8 @@ struct snd_sof_dsp_ops {
 	int (*get_bar_index)(struct snd_sof_dev *sdev,
 			     u32 type); /* optional */
 	int (*get_mailbox_offset)(struct snd_sof_dev *sdev);/* mandatory for common loader code */
+	int (*get_window_offset)(struct snd_sof_dev *sdev,
+				 u32 id);/* mandatory for common loader code */
 
 	/* DAI ops */
 	struct snd_soc_dai_driver *drv;
-- 
cgit v1.2.3-59-g8ed1b


From 83ee7ab1627b7597bbb9acc26b37e9f96fff40c9 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 7 Aug 2019 10:02:01 -0500
Subject: ASoC: SOF: Intel: byt: Refactor fw ready / mem windows creation

There is a lot of duplicate code when processing IPC firmware ready
notification and creating memory windows.

First step in reducing the code duplication is to introduce generic
functions:
	* sof_get_windows
	* sof_fw_ready
that will replace, in the first step, the specific implementation related
to baytrail related platforms:
	* byt_get_windows
	* byt_fw_ready

So we are basically moving code from intel/byt.c to loader.c keeping
in mind that mbox_offset is a per platform constant so we need to
use newly introduced snd_sof_dsp_get_mailbox_offset /
snd_sof_dsp_get_window_offset in order to get the correct
mbox offset / window offset value.

Also, bar is a per platform constant so we use snd_sof_dsp_get_bar_index
instead of the hardcoded BYT_DSP_BAR.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190807150203.26359-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/byt.c | 164 ++++++--------------------------------------
 sound/soc/sof/loader.c    | 168 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/sof/sof-priv.h  |   2 +
 3 files changed, 189 insertions(+), 145 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index cb8f02ee09f2..5e7a6aaa627a 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -109,148 +109,6 @@ static void byt_host_done(struct snd_sof_dev *sdev);
 static void byt_dsp_done(struct snd_sof_dev *sdev);
 static void byt_get_reply(struct snd_sof_dev *sdev);
 
-/*
- * IPC Firmware ready.
- */
-static void byt_get_windows(struct snd_sof_dev *sdev)
-{
-	struct sof_ipc_window_elem *elem;
-	u32 outbox_offset = 0;
-	u32 stream_offset = 0;
-	u32 inbox_offset = 0;
-	u32 outbox_size = 0;
-	u32 stream_size = 0;
-	u32 inbox_size = 0;
-	int i;
-
-	if (!sdev->info_window) {
-		dev_err(sdev->dev, "error: have no window info\n");
-		return;
-	}
-
-	for (i = 0; i < sdev->info_window->num_windows; i++) {
-		elem = &sdev->info_window->window[i];
-
-		switch (elem->type) {
-		case SOF_IPC_REGION_UPBOX:
-			inbox_offset = elem->offset + MBOX_OFFSET;
-			inbox_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BYT_DSP_BAR] +
-						inbox_offset,
-						elem->size, "inbox",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_DOWNBOX:
-			outbox_offset = elem->offset + MBOX_OFFSET;
-			outbox_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BYT_DSP_BAR] +
-						outbox_offset,
-						elem->size, "outbox",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_TRACE:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BYT_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "etrace",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_DEBUG:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BYT_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "debug",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_STREAM:
-			stream_offset = elem->offset + MBOX_OFFSET;
-			stream_size = elem->size;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BYT_DSP_BAR] +
-						stream_offset,
-						elem->size, "stream",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_REGS:
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BYT_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "regs",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		case SOF_IPC_REGION_EXCEPTION:
-			sdev->dsp_oops_offset = elem->offset + MBOX_OFFSET;
-			snd_sof_debugfs_io_item(sdev,
-						sdev->bar[BYT_DSP_BAR] +
-						elem->offset +
-						MBOX_OFFSET,
-						elem->size, "exception",
-						SOF_DEBUGFS_ACCESS_D0_ONLY);
-			break;
-		default:
-			dev_err(sdev->dev, "error: get illegal window info\n");
-			return;
-		}
-	}
-
-	if (outbox_size == 0 || inbox_size == 0) {
-		dev_err(sdev->dev, "error: get illegal mailbox window\n");
-		return;
-	}
-
-	snd_sof_dsp_mailbox_init(sdev, inbox_offset, inbox_size,
-				 outbox_offset, outbox_size);
-	sdev->stream_box.offset = stream_offset;
-	sdev->stream_box.size = stream_size;
-
-	dev_dbg(sdev->dev, " mailbox upstream 0x%x - size 0x%x\n",
-		inbox_offset, inbox_size);
-	dev_dbg(sdev->dev, " mailbox downstream 0x%x - size 0x%x\n",
-		outbox_offset, outbox_size);
-	dev_dbg(sdev->dev, " stream region 0x%x - size 0x%x\n",
-		stream_offset, stream_size);
-}
-
-/* check for ABI compatibility and create memory windows on first boot */
-static int byt_fw_ready(struct snd_sof_dev *sdev, u32 msg_id)
-{
-	struct sof_ipc_fw_ready *fw_ready = &sdev->fw_ready;
-	u32 offset;
-	int ret;
-
-	/* mailbox must be on 4k boundary */
-	offset = MBOX_OFFSET;
-
-	dev_dbg(sdev->dev, "ipc: DSP is ready 0x%8.8x offset 0x%x\n",
-		msg_id, offset);
-
-	/* no need to re-check version/ABI for subsequent boots */
-	if (!sdev->first_boot)
-		return 0;
-
-	/* copy data from the DSP FW ready offset */
-	sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready,
-		       sizeof(*fw_ready));
-
-	/* make sure ABI version is compatible */
-	ret = snd_sof_ipc_valid(sdev);
-	if (ret < 0)
-		return ret;
-
-	/* now check for extended data */
-	snd_sof_fw_parse_ext_data(sdev, sdev->mmio_bar, MBOX_OFFSET +
-				  sizeof(struct sof_ipc_fw_ready));
-
-	byt_get_windows(sdev);
-
-	return 0;
-}
-
 /*
  * Debug
  */
@@ -418,6 +276,16 @@ static void byt_get_reply(struct snd_sof_dev *sdev)
 	msg->reply_error = ret;
 }
 
+static int byt_get_mailbox_offset(struct snd_sof_dev *sdev)
+{
+	return MBOX_OFFSET;
+}
+
+static int byt_get_window_offset(struct snd_sof_dev *sdev, u32 id)
+{
+	return MBOX_OFFSET;
+}
+
 static void byt_host_done(struct snd_sof_dev *sdev)
 {
 	/* clear BUSY bit and set DONE bit - accept new messages */
@@ -612,7 +480,9 @@ const struct snd_sof_dsp_ops sof_tng_ops = {
 
 	/* ipc */
 	.send_msg	= byt_send_msg,
-	.fw_ready	= byt_fw_ready,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset = byt_get_mailbox_offset,
+	.get_window_offset = byt_get_window_offset,
 
 	.ipc_msg_data	= intel_ipc_msg_data,
 	.ipc_pcm_params	= intel_ipc_pcm_params,
@@ -771,7 +641,9 @@ const struct snd_sof_dsp_ops sof_byt_ops = {
 
 	/* ipc */
 	.send_msg	= byt_send_msg,
-	.fw_ready	= byt_fw_ready,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset = byt_get_mailbox_offset,
+	.get_window_offset = byt_get_window_offset,
 
 	.ipc_msg_data	= intel_ipc_msg_data,
 	.ipc_pcm_params	= intel_ipc_pcm_params,
@@ -828,7 +700,9 @@ const struct snd_sof_dsp_ops sof_cht_ops = {
 
 	/* ipc */
 	.send_msg	= byt_send_msg,
-	.fw_ready	= byt_fw_ready,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset = byt_get_mailbox_offset,
+	.get_window_offset = byt_get_window_offset,
 
 	.ipc_msg_data	= intel_ipc_msg_data,
 	.ipc_pcm_params	= intel_ipc_pcm_params,
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 93cb8fd0844f..d7f32745fefe 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -87,6 +87,174 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
 }
 EXPORT_SYMBOL(snd_sof_fw_parse_ext_data);
 
+/*
+ * IPC Firmware ready.
+ */
+static void sof_get_windows(struct snd_sof_dev *sdev)
+{
+	struct sof_ipc_window_elem *elem;
+	u32 outbox_offset = 0;
+	u32 stream_offset = 0;
+	u32 inbox_offset = 0;
+	u32 outbox_size = 0;
+	u32 stream_size = 0;
+	u32 inbox_size = 0;
+	int window_offset;
+	int bar;
+	int i;
+
+	if (!sdev->info_window) {
+		dev_err(sdev->dev, "error: have no window info\n");
+		return;
+	}
+
+	bar = snd_sof_dsp_get_bar_index(sdev, SOF_FW_BLK_TYPE_SRAM);
+	if (bar < 0) {
+		dev_err(sdev->dev, "error: have no bar mapping\n");
+		return;
+	}
+
+	for (i = 0; i < sdev->info_window->num_windows; i++) {
+		elem = &sdev->info_window->window[i];
+
+		window_offset = snd_sof_dsp_get_window_offset(sdev, elem->id);
+		if (window_offset < 0) {
+			dev_warn(sdev->dev, "warn: no offset for window %d\n",
+				 elem->id);
+			continue;
+		}
+
+		switch (elem->type) {
+		case SOF_IPC_REGION_UPBOX:
+			inbox_offset = window_offset + elem->offset;
+			inbox_size = elem->size;
+			snd_sof_debugfs_io_item(sdev,
+						sdev->bar[bar] +
+						inbox_offset,
+						elem->size, "inbox",
+						SOF_DEBUGFS_ACCESS_D0_ONLY);
+			break;
+		case SOF_IPC_REGION_DOWNBOX:
+			outbox_offset = window_offset + elem->offset;
+			outbox_size = elem->size;
+			snd_sof_debugfs_io_item(sdev,
+						sdev->bar[bar] +
+						outbox_offset,
+						elem->size, "outbox",
+						SOF_DEBUGFS_ACCESS_D0_ONLY);
+			break;
+		case SOF_IPC_REGION_TRACE:
+			snd_sof_debugfs_io_item(sdev,
+						sdev->bar[bar] +
+						window_offset +
+						elem->offset,
+						elem->size, "etrace",
+						SOF_DEBUGFS_ACCESS_D0_ONLY);
+			break;
+		case SOF_IPC_REGION_DEBUG:
+			snd_sof_debugfs_io_item(sdev,
+						sdev->bar[bar] +
+						window_offset +
+						elem->offset,
+						elem->size, "debug",
+						SOF_DEBUGFS_ACCESS_D0_ONLY);
+			break;
+		case SOF_IPC_REGION_STREAM:
+			stream_offset = window_offset + elem->offset;
+			stream_size = elem->size;
+			snd_sof_debugfs_io_item(sdev,
+						sdev->bar[bar] +
+						stream_offset,
+						elem->size, "stream",
+						SOF_DEBUGFS_ACCESS_D0_ONLY);
+			break;
+		case SOF_IPC_REGION_REGS:
+			snd_sof_debugfs_io_item(sdev,
+						sdev->bar[bar] +
+						window_offset +
+						elem->offset,
+						elem->size, "regs",
+						SOF_DEBUGFS_ACCESS_D0_ONLY);
+			break;
+		case SOF_IPC_REGION_EXCEPTION:
+			sdev->dsp_oops_offset = window_offset + elem->offset;
+			snd_sof_debugfs_io_item(sdev,
+						sdev->bar[bar] +
+						window_offset +
+						elem->offset,
+						elem->size, "exception",
+						SOF_DEBUGFS_ACCESS_D0_ONLY);
+			break;
+		default:
+			dev_err(sdev->dev, "error: get illegal window info\n");
+			return;
+		}
+	}
+
+	if (outbox_size == 0 || inbox_size == 0) {
+		dev_err(sdev->dev, "error: get illegal mailbox window\n");
+		return;
+	}
+
+	snd_sof_dsp_mailbox_init(sdev, inbox_offset, inbox_size,
+				 outbox_offset, outbox_size);
+	sdev->stream_box.offset = stream_offset;
+	sdev->stream_box.size = stream_size;
+
+	dev_dbg(sdev->dev, " mailbox upstream 0x%x - size 0x%x\n",
+		inbox_offset, inbox_size);
+	dev_dbg(sdev->dev, " mailbox downstream 0x%x - size 0x%x\n",
+		outbox_offset, outbox_size);
+	dev_dbg(sdev->dev, " stream region 0x%x - size 0x%x\n",
+		stream_offset, stream_size);
+}
+
+/* check for ABI compatibility and create memory windows on first boot */
+int sof_fw_ready(struct snd_sof_dev *sdev, u32 msg_id)
+{
+	struct sof_ipc_fw_ready *fw_ready = &sdev->fw_ready;
+	int offset;
+	int bar;
+	int ret;
+
+	/* mailbox must be on 4k boundary */
+	offset = snd_sof_dsp_get_mailbox_offset(sdev);
+	if (offset < 0) {
+		dev_err(sdev->dev, "error: have no mailbox offset\n");
+		return offset;
+	}
+
+	bar = snd_sof_dsp_get_bar_index(sdev, SOF_FW_BLK_TYPE_SRAM);
+	if (bar < 0) {
+		dev_err(sdev->dev, "error: have no bar mapping\n");
+		return -EINVAL;
+	}
+
+	dev_dbg(sdev->dev, "ipc: DSP is ready 0x%8.8x offset 0x%x\n",
+		msg_id, offset);
+
+	/* no need to re-check version/ABI for subsequent boots */
+	if (!sdev->first_boot)
+		return 0;
+
+	/* copy data from the DSP FW ready offset */
+	sof_block_read(sdev, bar, offset, fw_ready, sizeof(*fw_ready));
+
+	/* make sure ABI version is compatible */
+	ret = snd_sof_ipc_valid(sdev);
+	if (ret < 0)
+		return ret;
+
+	/* now check for extended data */
+	snd_sof_fw_parse_ext_data(sdev, bar, offset +
+				  sizeof(struct sof_ipc_fw_ready));
+
+	sof_get_windows(sdev);
+
+	return 0;
+}
+EXPORT_SYMBOL(sof_fw_ready);
+
 /* generic module parser for mmaped DSPs */
 int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev,
 				struct snd_sof_mod_hdr *module)
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index de11f28bae3a..730f3259dd02 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -642,6 +642,8 @@ void sof_block_write(struct snd_sof_dev *sdev, u32 bar, u32 offset, void *src,
 void sof_block_read(struct snd_sof_dev *sdev, u32 bar, u32 offset, void *dest,
 		    size_t size);
 
+int sof_fw_ready(struct snd_sof_dev *sdev, u32 msg_id);
+
 void intel_ipc_msg_data(struct snd_sof_dev *sdev,
 			struct snd_pcm_substream *substream,
 			void *p, size_t sz);
-- 
cgit v1.2.3-59-g8ed1b


From 40073a9d53eac3f3ebf569aa9afd61301eff9190 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Fri, 9 Aug 2019 19:01:00 +0800
Subject: ASoC: SOF: Intel: Add missing include file hdac_hda.h

Building with SND_SOC_SOF_HDA_AUDIO_CODEC fails:

sound/soc/sof/intel/hda-bus.c: In function sof_hda_bus_init:
sound/soc/sof/intel/hda-bus.c:16:25: error: implicit declaration of function
 snd_soc_hdac_hda_get_ops; did you mean snd_soc_jack_add_gpiods? [-Werror=implicit-function-declaration]
 #define sof_hda_ext_ops snd_soc_hdac_hda_get_ops()

Reported-by: Hulk Robot <hulkci@huawei.com>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Fixes: d4ff1b3917a5 ("ASoC: SOF: Intel: Initialize hdaudio bus properly")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/sof/intel/hda-bus.c | 1 +
 sound/soc/sof/intel/hda.c     | 3 ---
 2 files changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c
index 0caec3a070d3..1d2babdda9dd 100644
--- a/sound/soc/sof/intel/hda-bus.c
+++ b/sound/soc/sof/intel/hda-bus.c
@@ -13,6 +13,7 @@
 #include "hda.h"
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+#include "../../codecs/hdac_hda.h"
 #define sof_hda_ext_ops	snd_soc_hdac_hda_get_ops()
 #else
 #define sof_hda_ext_ops	NULL
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 7ca27000c34d..dd6c8ad62b3e 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -23,9 +23,6 @@
 #include <sound/sof/xtensa.h>
 #include "../ops.h"
 #include "hda.h"
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
-#include "../../codecs/hdac_hda.h"
-#endif
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 #include <sound/soc-acpi-intel-match.h>
-- 
cgit v1.2.3-59-g8ed1b


From f43ec25aa5480515a39a405084ca8cb5e2c408aa Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 9 Aug 2019 16:13:16 +0200
Subject: ALSA: hda - Inform too slow responses

For the sake of debugging, show each response that took too long time
and its corresponding last verb.  Since such an error happens in
series once when started, use the ratelimited variant for suppressing
the flood.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_controller.c | 13 +++++++++++--
 1 file changed, 11 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 156930d0c9d3..ec0608d44263 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -789,6 +789,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
 	unsigned long timeout;
 	unsigned long loopcounter;
 	int do_poll = 0;
+	bool warned = false;
 
  again:
 	timeout = jiffies + msecs_to_jiffies(1000);
@@ -808,9 +809,17 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
 		spin_unlock_irq(&bus->reg_lock);
 		if (time_after(jiffies, timeout))
 			break;
-		if (hbus->needs_damn_long_delay || loopcounter > 3000)
+#define LOOP_COUNT_MAX	3000
+		if (hbus->needs_damn_long_delay ||
+		    loopcounter > LOOP_COUNT_MAX) {
+			if (loopcounter > LOOP_COUNT_MAX && !warned) {
+				dev_dbg_ratelimited(chip->card->dev,
+						    "too slow response, last cmd=%#08x\n",
+						    bus->last_cmd[addr]);
+				warned = true;
+			}
 			msleep(2); /* temporary workaround */
-		else {
+		} else {
 			udelay(10);
 			cond_resched();
 		}
-- 
cgit v1.2.3-59-g8ed1b


From a2ef03fe617a8365fb7794531b11ba587509a9b9 Mon Sep 17 00:00:00 2001
From: Tomas Espeleta <tomas.espeleta@gmail.com>
Date: Fri, 9 Aug 2019 16:37:54 +0200
Subject: ALSA: hda - Add a quirk model for fixing Huawei Matebook X right
 speaker

[ This is rather a revival of the patch Tomas sent in months ago, but
  applying only with the quirk model option -- tiwai ]

Hard coded coefficients to make Huawuei Matebook X right speaker
work. The Matebook X has a ALC298, please refer to bug 197801 on
how these numbers were reverse engineered from the Windows driver

The reversed engineered sequence represents a repeating pattern
of verbs, and the only values that are changing periodically are
written on indexes 0x23 and 0x25:

0x500, 0x23
0x400, VALUE1
0x500, 0x25
0x400, VALUE2

* skipped reading sequences (0x500 - 0xc00 sequences are ignored)
* static values from reverse engineering are used

NOTE: since a significant risk is still considered, this is provided
as an experimental fix that isn't applied as default for now.  For
enabling the fix, you'll have to choose huawei-mbx-stereo via model
option of snd-hda-intel module.

If we get feedback from users that this works stably, we may apply it
per default.

[ Some coding style fixes and replacement with AC_VERB_* by tiwai ]

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=197801
Signed-off-by: Tomas Espeleta <tomas.espeleta@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/hd-audio/models.rst |  3 ++
 sound/pci/hda/patch_realtek.c           | 74 +++++++++++++++++++++++++++++++++
 2 files changed, 77 insertions(+)

(limited to 'sound')

diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst
index 7d7c191102a7..11298f0ce44d 100644
--- a/Documentation/sound/hd-audio/models.rst
+++ b/Documentation/sound/hd-audio/models.rst
@@ -260,6 +260,9 @@ alc295-hp-x360
     HP Spectre X360 fixups
 alc-sense-combo
     Headset button support for Chrome platform
+huawei-mbx-stereo
+    Enable initialization verbs for Huawei MBX stereo speakers;
+    might be risky, try this at your own risk
 
 ALC66x/67x/892
 ==============
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index de224cbea7a0..cb7baa65b298 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3762,6 +3762,72 @@ static void alc269_x101_hp_automute_hook(struct hda_codec *codec,
 			    vref);
 }
 
+/*
+ * Magic sequence to make Huawei Matebook X right speaker working (bko#197801)
+ */
+struct hda_alc298_mbxinit {
+	unsigned char value_0x23;
+	unsigned char value_0x25;
+};
+
+static void alc298_huawei_mbx_stereo_seq(struct hda_codec *codec,
+					 const struct hda_alc298_mbxinit *initval,
+					 bool first)
+{
+	snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x0);
+	alc_write_coef_idx(codec, 0x26, 0xb000);
+
+	if (first)
+		snd_hda_codec_write(codec, 0x21, 0, AC_VERB_GET_PIN_SENSE, 0x0);
+
+	snd_hda_codec_write(codec, 0x6, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80);
+	alc_write_coef_idx(codec, 0x26, 0xf000);
+	alc_write_coef_idx(codec, 0x23, initval->value_0x23);
+
+	if (initval->value_0x23 != 0x1e)
+		alc_write_coef_idx(codec, 0x25, initval->value_0x25);
+
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010);
+}
+
+static void alc298_fixup_huawei_mbx_stereo(struct hda_codec *codec,
+					   const struct hda_fixup *fix,
+					   int action)
+{
+	/* Initialization magic */
+	static const struct hda_alc298_mbxinit dac_init[] = {
+		{0x0c, 0x00}, {0x0d, 0x00}, {0x0e, 0x00}, {0x0f, 0x00},
+		{0x10, 0x00}, {0x1a, 0x40}, {0x1b, 0x82}, {0x1c, 0x00},
+		{0x1d, 0x00}, {0x1e, 0x00}, {0x1f, 0x00},
+		{0x20, 0xc2}, {0x21, 0xc8}, {0x22, 0x26}, {0x23, 0x24},
+		{0x27, 0xff}, {0x28, 0xff}, {0x29, 0xff}, {0x2a, 0x8f},
+		{0x2b, 0x02}, {0x2c, 0x48}, {0x2d, 0x34}, {0x2e, 0x00},
+		{0x2f, 0x00},
+		{0x30, 0x00}, {0x31, 0x00}, {0x32, 0x00}, {0x33, 0x00},
+		{0x34, 0x00}, {0x35, 0x01}, {0x36, 0x93}, {0x37, 0x0c},
+		{0x38, 0x00}, {0x39, 0x00}, {0x3a, 0xf8}, {0x38, 0x80},
+		{}
+	};
+	const struct hda_alc298_mbxinit *seq;
+
+	if (action != HDA_FIXUP_ACT_INIT)
+		return;
+
+	/* Start */
+	snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x00);
+	snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80);
+	alc_write_coef_idx(codec, 0x26, 0xf000);
+	alc_write_coef_idx(codec, 0x22, 0x31);
+	alc_write_coef_idx(codec, 0x23, 0x0b);
+	alc_write_coef_idx(codec, 0x25, 0x00);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010);
+
+	for (seq = dac_init; seq->value_0x23; seq++)
+		alc298_huawei_mbx_stereo_seq(codec, seq, seq == dac_init);
+}
+
 static void alc269_fixup_x101_headset_mic(struct hda_codec *codec,
 				     const struct hda_fixup *fix, int action)
 {
@@ -5787,6 +5853,7 @@ enum {
 	ALC255_FIXUP_DUMMY_LINEOUT_VERB,
 	ALC255_FIXUP_DELL_HEADSET_MIC,
 	ALC256_FIXUP_HUAWEI_MACH_WX9_PINS,
+	ALC298_FIXUP_HUAWEI_MBX_STEREO,
 	ALC295_FIXUP_HP_X360,
 	ALC221_FIXUP_HP_HEADSET_MIC,
 	ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
@@ -6095,6 +6162,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_MIC_MUTE_LED
 	},
+	[ALC298_FIXUP_HUAWEI_MBX_STEREO] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc298_fixup_huawei_mbx_stereo,
+		.chained = true,
+		.chain_id = ALC255_FIXUP_MIC_MUTE_LED
+	},
 	[ALC269_FIXUP_ASUS_X101_FUNC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_x101_headset_mic,
@@ -7272,6 +7345,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"},
 	{.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"},
 	{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
+	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
-- 
cgit v1.2.3-59-g8ed1b


From 332ccf00bf85adbf48015084be0e60f5cc57a055 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 9 Aug 2019 17:15:31 +0200
Subject: ASoC: hdac_hdmi: Offload dapm update at jack detection

hdac_hdmi_present_sense() calls the audio component to get ELD update,
then it reports the jack status change and updates DAPM graph
accordingly.  This works when it's called from the normal code paths.

However, it may lead to a dead lock when it's called from the audio
component notifier.  Namely, the DAPM update involves with the runtime
PM, and it eventually calls again the audio component get_power()
ops.  Since i915 driver already takes a mutex around the audio
component ops calls, we'll eventually get the mutex doubly.

As a workaround, in this patch, only the jack state is updated in the
code path from hdac_hdmi_eld_notify_cb(), and the DAPM update is
deferred to a work so that it's processed in another context.

Reported-by: Imre Deak <imre.deak@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20190809151531.24359-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdac_hdmi.c | 57 +++++++++++++++++++++++++++++++++++++-------
 1 file changed, 48 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 47eee18b66a3..11ec031ad749 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -88,8 +88,10 @@ struct hdac_hdmi_port {
 	hda_nid_t mux_nids[HDA_MAX_CONNECTIONS];
 	struct hdac_hdmi_eld eld;
 	const char *jack_pin;
+	bool is_connect;
 	struct snd_soc_dapm_context *dapm;
 	const char *output_pin;
+	struct work_struct dapm_work;
 };
 
 struct hdac_hdmi_pcm {
@@ -163,11 +165,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
 {
 	struct hdac_device *hdev = port->pin->hdev;
 
-	if (is_connect)
-		snd_soc_dapm_enable_pin(port->dapm, port->jack_pin);
-	else
-		snd_soc_dapm_disable_pin(port->dapm, port->jack_pin);
-
+	port->is_connect = is_connect;
 	if (is_connect) {
 		/*
 		 * Report Jack connect event when a device is connected
@@ -193,10 +191,32 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
 		if (pcm->jack_event > 0)
 			pcm->jack_event--;
 	}
+}
 
+static void hdac_hdmi_port_dapm_update(struct hdac_hdmi_port *port)
+{
+	if (port->is_connect)
+		snd_soc_dapm_enable_pin(port->dapm, port->jack_pin);
+	else
+		snd_soc_dapm_disable_pin(port->dapm, port->jack_pin);
 	snd_soc_dapm_sync(port->dapm);
 }
 
+static void hdac_hdmi_jack_dapm_work(struct work_struct *work)
+{
+	struct hdac_hdmi_port *port;
+
+	port = container_of(work, struct hdac_hdmi_port, dapm_work);
+	hdac_hdmi_port_dapm_update(port);
+}
+
+static void hdac_hdmi_jack_report_sync(struct hdac_hdmi_pcm *pcm,
+		struct hdac_hdmi_port *port, bool is_connect)
+{
+	hdac_hdmi_jack_report(pcm, port, is_connect);
+	hdac_hdmi_port_dapm_update(port);
+}
+
 /* MST supported verbs */
 /*
  * Get the no devices that can be connected to a port on the Pin widget.
@@ -873,7 +893,7 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol,
 		list_for_each_entry_safe(p, p_next, &pcm->port_list, head) {
 			if (p == port && p->id == port->id &&
 					p->pin == port->pin) {
-				hdac_hdmi_jack_report(pcm, port, false);
+				hdac_hdmi_jack_report_sync(pcm, port, false);
 				list_del(&p->head);
 			}
 		}
@@ -887,7 +907,7 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol,
 		if (!strcmp(cvt_name, pcm->cvt->name)) {
 			list_add_tail(&port->head, &pcm->port_list);
 			if (port->eld.monitor_present && port->eld.eld_valid) {
-				hdac_hdmi_jack_report(pcm, port, true);
+				hdac_hdmi_jack_report_sync(pcm, port, true);
 				mutex_unlock(&hdmi->pin_mutex);
 				return ret;
 			}
@@ -1250,16 +1270,20 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin,
 		 * report jack here. It will be done in usermode mux
 		 * control select.
 		 */
-		if (pcm)
+		if (pcm) {
 			hdac_hdmi_jack_report(pcm, port, false);
+			schedule_work(&port->dapm_work);
+		}
 
 		mutex_unlock(&hdmi->pin_mutex);
 		return;
 	}
 
 	if (port->eld.monitor_present && port->eld.eld_valid) {
-		if (pcm)
+		if (pcm) {
 			hdac_hdmi_jack_report(pcm, port, true);
+			schedule_work(&port->dapm_work);
+		}
 
 		print_hex_dump_debug("ELD: ", DUMP_PREFIX_OFFSET, 16, 1,
 			  port->eld.eld_buffer, port->eld.eld_size, false);
@@ -1288,6 +1312,7 @@ static int hdac_hdmi_add_ports(struct hdac_device *hdev,
 	for (i = 0; i < max_ports; i++) {
 		ports[i].id = i;
 		ports[i].pin = pin;
+		INIT_WORK(&ports[i].dapm_work, hdac_hdmi_jack_dapm_work);
 	}
 	pin->ports = ports;
 	pin->num_ports = max_ports;
@@ -2052,8 +2077,20 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev)
 	return ret;
 }
 
+static void clear_dapm_works(struct hdac_device *hdev)
+{
+	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
+	struct hdac_hdmi_pin *pin;
+	int i;
+
+	list_for_each_entry(pin, &hdmi->pin_list, head)
+		for (i = 0; i < pin->num_ports; i++)
+			cancel_work_sync(&pin->ports[i].dapm_work);
+}
+
 static int hdac_hdmi_dev_remove(struct hdac_device *hdev)
 {
+	clear_dapm_works(hdev);
 	snd_hdac_display_power(hdev->bus, hdev->addr, false);
 
 	return 0;
@@ -2072,6 +2109,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 	if (!bus)
 		return 0;
 
+	clear_dapm_works(hdev);
+
 	/*
 	 * Power down afg.
 	 * codec_read is preferred over codec_write to set the power state.
-- 
cgit v1.2.3-59-g8ed1b


From 3dc29b8b2062075602c7aff1514a120b4ed0187f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:52:33 +0900
Subject: ASoC: soc-core: support snd_soc_dai_link_component for aux_dev

To find aux_dev, ASoC is using .name, codec_name, codec_of_node.
Here, .name is used to fallback in case of no codec.

But, we already have this kind of component finding method by
snd_soc_dai_link_component and soc_find_component().
We shouldn't have duplicated implementation to do same things.
This patch adds snd_soc_dai_link_component support to finding aux_dev.

Now, no driver is using only .name.
All drivers are using codec_name and/or codec_of_node.
This means no driver is finding component from .name so far.
(Actually almost all drivers are using .name as just "device name",
 not for finding component...)

This patch
1) add snd_soc_dai_link_component support for aux_dev. legacy style will
   be removed if all drivers are switched to new style.
2) try to find component via snd_soc_dai_link_component.
   Then, it doesn't try to find via .name, because no driver is using
   it so far.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y3046wcf.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  |  7 +++++++
 sound/soc/soc-core.c | 36 ++++++++++--------------------------
 2 files changed, 17 insertions(+), 26 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 85ad971e9432..fd6ecea48fc0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -941,6 +941,7 @@ struct snd_soc_dai_link {
 #define COMP_CPU(_dai)			{ .dai_name = _dai, }
 #define COMP_CODEC(_name, _dai)		{ .name = _name, .dai_name = _dai, }
 #define COMP_PLATFORM(_name)		{ .name = _name }
+#define COMP_AUX(_name)			{ .name = _name }
 #define COMP_DUMMY()			{ .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", }
 
 extern struct snd_soc_dai_link_component null_dailink_component[0];
@@ -971,6 +972,12 @@ struct snd_soc_aux_dev {
 	const char *codec_name;
 	struct device_node *codec_of_node;
 
+	/*
+	 * name, codec_name, codec_of_node will be replaced
+	 * into dlc. don't use both in the same time
+	 */
+	struct snd_soc_dai_link_component dlc;
+
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_component *component);
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bf45e60eb34f..56b99e340dda 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1537,38 +1537,22 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
 {
 	struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
 	struct snd_soc_component *component;
-	struct snd_soc_dai_link_component dlc;
 
-	if (aux_dev->codec_of_node || aux_dev->codec_name) {
-		/* codecs, usually analog devices */
-		dlc.name = aux_dev->codec_name;
-		dlc.of_node = aux_dev->codec_of_node;
-		component = soc_find_component(&dlc);
-		if (!component) {
-			if (dlc.of_node)
-				dlc.name = of_node_full_name(dlc.of_node);
-			goto err_defer;
-		}
-	} else if (aux_dev->name) {
-		/* generic components */
-		dlc.name = aux_dev->name;
-		dlc.of_node = NULL;
-		component = soc_find_component(&dlc);
-		if (!component)
-			goto err_defer;
-	} else {
-		dev_err(card->dev, "ASoC: Invalid auxiliary device\n");
-		return -EINVAL;
-	}
+	/* remove me */
+	if (aux_dev->codec_name)
+		aux_dev->dlc.name = aux_dev->codec_name;
+	if (aux_dev->codec_of_node)
+		aux_dev->dlc.of_node = aux_dev->codec_of_node;
+
+	/* codecs, usually analog devices */
+	component = soc_find_component(&aux_dev->dlc);
+	if (!component)
+		return -EPROBE_DEFER;
 
 	component->init = aux_dev->init;
 	list_add(&component->card_aux_list, &card->aux_comp_list);
 
 	return 0;
-
-err_defer:
-	dev_err(card->dev, "ASoC: %s not registered\n", dlc.name);
-	return -EPROBE_DEFER;
 }
 
 static int soc_probe_aux_devices(struct snd_soc_card *card)
-- 
cgit v1.2.3-59-g8ed1b


From 14fdfc058d812f683d9aa51083d1a920aa6a1eb9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:52:47 +0900
Subject: ASoC: simple-card: use snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wofo6wc1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index ef849151ba56..0eac3bcb9736 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -424,7 +424,7 @@ static int simple_parse_aux_devs(struct device_node *node,
 		aux_node = of_parse_phandle(node, PREFIX "aux-devs", i);
 		if (!aux_node)
 			return -EINVAL;
-		card->aux_dev[i].codec_of_node = aux_node;
+		card->aux_dev[i].dlc.of_node = aux_node;
 	}
 
 	card->num_aux_devs = n;
-- 
cgit v1.2.3-59-g8ed1b


From aa1afd92f374a895ab74cb96dacd25a3485a08c4 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:53:33 +0900
Subject: ASoC: intel: cht_bsw_max98090_ti: use snd_soc_dai_link_component for
 aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v9v86war.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 33eb72545be6..1db9a95e6a79 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -324,9 +324,8 @@ static const struct snd_soc_ops cht_be_ssp2_ops = {
 };
 
 static struct snd_soc_aux_dev cht_max98090_headset_dev = {
-	.name = "Headset Chip",
+	.dlc = COMP_AUX("i2c-104C227E:00"),
 	.init = cht_max98090_headset_init,
-	.codec_name = "i2c-104C227E:00",
 };
 
 SND_SOC_DAILINK_DEF(dummy,
-- 
cgit v1.2.3-59-g8ed1b


From 5c9e38cb57ce07d780dc05d34ae140271c2da159 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:53:45 +0900
Subject: ASoC: mediatek: mt8183-da7219-max98357: use
 snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tvas6waf.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index 59076e21cb47..2a6097174614 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -370,7 +370,7 @@ static int
 mt8183_da7219_max98357_headset_init(struct snd_soc_component *component);
 
 static struct snd_soc_aux_dev mt8183_da7219_max98357_headset_dev = {
-	.name = "Headset Chip",
+	.dlc = COMP_EMPTY(),
 	.init = mt8183_da7219_max98357_headset_init,
 };
 
@@ -436,10 +436,10 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev)
 		dai_link->platforms->of_node = platform_node;
 	}
 
-	mt8183_da7219_max98357_headset_dev.codec_of_node =
+	mt8183_da7219_max98357_headset_dev.dlc.of_node =
 		of_parse_phandle(pdev->dev.of_node,
 				 "mediatek,headset-codec", 0);
-	if (!mt8183_da7219_max98357_headset_dev.codec_of_node) {
+	if (!mt8183_da7219_max98357_headset_dev.dlc.of_node) {
 		dev_err(&pdev->dev,
 			"Property 'mediatek,headset-codec' missing/invalid\n");
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From a73b522d4b570ee41754990e81f6f4e849af1af9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:53:59 +0900
Subject: ASoC: meson: axg-card: use snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r25w6wa1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-card.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 14a8321744da..6283e5025548 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -124,7 +124,7 @@ static void axg_card_clean_references(struct axg_card *priv)
 
 	if (card->aux_dev) {
 		for (i = 0; i < card->num_aux_devs; i++)
-			of_node_put(card->aux_dev[i].codec_of_node);
+			of_node_put(card->aux_dev[i].dlc.of_node);
 	}
 
 	kfree(card->dai_link);
@@ -158,9 +158,9 @@ static int axg_card_add_aux_devices(struct snd_soc_card *card)
 	card->num_aux_devs = num;
 
 	for (i = 0; i < card->num_aux_devs; i++, aux++) {
-		aux->codec_of_node =
+		aux->dlc.of_node =
 			of_parse_phandle(node, "audio-aux-devs", i);
-		if (!aux->codec_of_node)
+		if (!aux->dlc.of_node)
 			return -EINVAL;
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From 70a6b7bfba9bfd781b8c137f059487017dd385d8 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:05 +0900
Subject: ASoC: rockchip: rockchip_max98090: use snd_soc_dai_link_component for
 aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pnlg6w9v.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_max98090.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c
index 1af1147c3da3..7b0c21fa6dca 100644
--- a/sound/soc/rockchip/rockchip_max98090.c
+++ b/sound/soc/rockchip/rockchip_max98090.c
@@ -160,7 +160,7 @@ static struct snd_soc_dai_link rk_dailink = {
 static int rk_98090_headset_init(struct snd_soc_component *component);
 
 static struct snd_soc_aux_dev rk_98090_headset_dev = {
-	.name = "Headset Chip",
+	.dlc = COMP_EMPTY(),
 	.init = rk_98090_headset_init,
 };
 
@@ -226,9 +226,9 @@ static int snd_rk_mc_probe(struct platform_device *pdev)
 
 	rk_dailink.platforms->of_node = rk_dailink.cpus->of_node;
 
-	rk_98090_headset_dev.codec_of_node = of_parse_phandle(np,
+	rk_98090_headset_dev.dlc.of_node = of_parse_phandle(np,
 			"rockchip,headset-codec", 0);
-	if (!rk_98090_headset_dev.codec_of_node) {
+	if (!rk_98090_headset_dev.dlc.of_node) {
 		dev_err(&pdev->dev,
 			"Property 'rockchip,headset-codec' missing/invalid\n");
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From d20bb789d7283bcbacd3b12c60668c473bc1dac0 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:10 +0900
Subject: ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for
 aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/87o9106w9p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/neo1973_wm8753.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 396776ffd670..38f536bafa09 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -297,8 +297,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 
 static struct snd_soc_aux_dev neo1973_aux_devs[] = {
 	{
-		.name = "dfbmcs320",
-		.codec_name = "dfbmcs320.0",
+		.dlc = COMP_AUX("dfbmcs320.0"),
 	},
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 2d946aaa80c79452c700381b4c1f06f11dfd2bdd Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:15 +0900
Subject: ASoC: samsung: speyside: use snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/87mugk6w9l.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/speyside.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 51e4c976c8be..9e58cbed942a 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -240,8 +240,7 @@ static int speyside_wm9081_init(struct snd_soc_component *component)
 
 static struct snd_soc_aux_dev speyside_aux_dev[] = {
 	{
-		.name = "wm9081",
-		.codec_name = "wm9081.1-006c",
+		.dlc = COMP_AUX("wm9081.1-006c"),
 		.init = speyside_wm9081_init,
 	},
 };
-- 
cgit v1.2.3-59-g8ed1b


From 124749194ad2e08fbea9c34fa5e2728310bf4486 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:20 +0900
Subject: ASoC: samsung: tm2_wm5110: use snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/87lfw46w9g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/tm2_wm5110.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index c091033d17ad..bb9910d4cbe2 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -307,7 +307,6 @@ static struct snd_soc_aux_dev tm2_speaker_amp_dev;
 static int tm2_late_probe(struct snd_soc_card *card)
 {
 	struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
-	struct snd_soc_dai_link_component dlc = { 0 };
 	unsigned int ch_map[] = { 0, 1 };
 	struct snd_soc_dai *amp_pdm_dai;
 	struct snd_soc_pcm_runtime *rtd;
@@ -334,8 +333,7 @@ static int tm2_late_probe(struct snd_soc_card *card)
 		return ret;
 	}
 
-	dlc.of_node = tm2_speaker_amp_dev.codec_of_node;
-	amp_pdm_dai = snd_soc_find_dai(&dlc);
+	amp_pdm_dai = snd_soc_find_dai(&tm2_speaker_amp_dev.dlc);
 	if (!amp_pdm_dai)
 		return -ENODEV;
 
@@ -532,9 +530,9 @@ static int tm2_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	card->aux_dev[0].codec_of_node = of_parse_phandle(dev->of_node,
+	card->aux_dev[0].dlc.of_node = of_parse_phandle(dev->of_node,
 							"audio-amplifier", 0);
-	if (!card->aux_dev[0].codec_of_node) {
+	if (!card->aux_dev[0].dlc.of_node) {
 		dev_err(dev, "audio-amplifier property invalid or missing\n");
 		return -EINVAL;
 	}
@@ -623,7 +621,7 @@ dai_node_put:
 		of_node_put(cpu_dai_node[i]);
 	}
 
-	of_node_put(card->aux_dev[0].codec_of_node);
+	of_node_put(card->aux_dev[0].dlc.of_node);
 
 	return ret;
 }
-- 
cgit v1.2.3-59-g8ed1b


From 3d0d2d64b7eb4f2a451fa184829b749851c14f55 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:25 +0900
Subject: ASoC: sunxi: sun4i-codec: use snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k1bo6w9b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-codec.c | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 619073e7d972..ee448d5e07a6 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -1424,7 +1424,7 @@ static const struct snd_soc_dapm_route sun8i_codec_card_routes[] = {
 };
 
 static struct snd_soc_aux_dev aux_dev = {
-	.name = "Codec Analog Controls",
+	.dlc = COMP_EMPTY(),
 };
 
 static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev)
@@ -1436,10 +1436,10 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev)
 	if (!card)
 		return ERR_PTR(-ENOMEM);
 
-	aux_dev.codec_of_node = of_parse_phandle(dev->of_node,
+	aux_dev.dlc.of_node = of_parse_phandle(dev->of_node,
 						 "allwinner,codec-analog-controls",
 						 0);
-	if (!aux_dev.codec_of_node) {
+	if (!aux_dev.dlc.of_node) {
 		dev_err(dev, "Can't find analog controls for codec.\n");
 		return ERR_PTR(-EINVAL);
 	};
@@ -1474,10 +1474,10 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev)
 	if (!card)
 		return ERR_PTR(-ENOMEM);
 
-	aux_dev.codec_of_node = of_parse_phandle(dev->of_node,
+	aux_dev.dlc.of_node = of_parse_phandle(dev->of_node,
 						 "allwinner,codec-analog-controls",
 						 0);
-	if (!aux_dev.codec_of_node) {
+	if (!aux_dev.dlc.of_node) {
 		dev_err(dev, "Can't find analog controls for codec.\n");
 		return ERR_PTR(-EINVAL);
 	};
@@ -1512,10 +1512,10 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev)
 	if (!card)
 		return ERR_PTR(-ENOMEM);
 
-	aux_dev.codec_of_node = of_parse_phandle(dev->of_node,
+	aux_dev.dlc.of_node = of_parse_phandle(dev->of_node,
 						 "allwinner,codec-analog-controls",
 						 0);
-	if (!aux_dev.codec_of_node) {
+	if (!aux_dev.dlc.of_node) {
 		dev_err(dev, "Can't find analog controls for codec.\n");
 		return ERR_PTR(-EINVAL);
 	};
-- 
cgit v1.2.3-59-g8ed1b


From 77b21d2822fdbfad19170516256c4c7f236054a9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:30 +0900
Subject: ASoC: ti: rx51: use snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87imr86w96.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/rx51.c | 14 ++++++--------
 1 file changed, 6 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index ccd0e8a07dd1..588f680a9c24 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -319,12 +319,10 @@ static struct snd_soc_dai_link rx51_dai[] = {
 
 static struct snd_soc_aux_dev rx51_aux_dev[] = {
 	{
-		.name = "TLV320AIC34b",
-		.codec_name = "tlv320aic3x-codec.2-0019",
+		.dlc = COMP_AUX("tlv320aic3x-codec.2-0019"),
 	},
 	{
-		.name = "TPA61320A2",
-		.codec_name = "tpa6130a2.2-0060",
+		.dlc = COMP_AUX("tpa6130a2.2-0060"),
 	},
 };
 
@@ -397,8 +395,8 @@ static int rx51_soc_probe(struct platform_device *pdev)
 			dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n");
 			return -EINVAL;
 		}
-		rx51_aux_dev[0].codec_name = NULL;
-		rx51_aux_dev[0].codec_of_node = dai_node;
+		rx51_aux_dev[0].dlc.name = NULL;
+		rx51_aux_dev[0].dlc.of_node = dai_node;
 		rx51_codec_conf[0].dev_name = NULL;
 		rx51_codec_conf[0].of_node = dai_node;
 
@@ -407,8 +405,8 @@ static int rx51_soc_probe(struct platform_device *pdev)
 			dev_err(&pdev->dev, "Headphone amplifier node is not provided\n");
 			return -EINVAL;
 		}
-		rx51_aux_dev[1].codec_name = NULL;
-		rx51_aux_dev[1].codec_of_node = dai_node;
+		rx51_aux_dev[1].dlc.name = NULL;
+		rx51_aux_dev[1].dlc.of_node = dai_node;
 		rx51_codec_conf[1].dev_name = NULL;
 		rx51_codec_conf[1].of_node = dai_node;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From a48b561d873d1d9fda55782d275eff94ec647863 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:39 +0900
Subject: ASoC: soc-core: remove legacy style of aux_dev

Now all drivers are using snd_soc_dai_link_component for aux_dev.
Let's remove legacy style

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h86s6w8x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 9 ---------
 sound/soc/soc-core.c | 6 ------
 2 files changed, 15 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index fd6ecea48fc0..2fc56e5963f3 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -963,19 +963,10 @@ struct snd_soc_codec_conf {
 };
 
 struct snd_soc_aux_dev {
-	const char *name;		/* Codec name */
-
 	/*
 	 * specify multi-codec either by device name, or by
 	 * DT/OF node, but not both.
 	 */
-	const char *codec_name;
-	struct device_node *codec_of_node;
-
-	/*
-	 * name, codec_name, codec_of_node will be replaced
-	 * into dlc. don't use both in the same time
-	 */
 	struct snd_soc_dai_link_component dlc;
 
 	/* codec/machine specific init - e.g. add machine controls */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 56b99e340dda..4af382d52675 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1538,12 +1538,6 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
 	struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
 	struct snd_soc_component *component;
 
-	/* remove me */
-	if (aux_dev->codec_name)
-		aux_dev->dlc.name = aux_dev->codec_name;
-	if (aux_dev->codec_of_node)
-		aux_dev->dlc.of_node = aux_dev->codec_of_node;
-
 	/* codecs, usually analog devices */
 	component = soc_find_component(&aux_dev->dlc);
 	if (!component)
-- 
cgit v1.2.3-59-g8ed1b


From dbf0649f4340c8bb7d36b8d6255dba03ed6981e7 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 8 Aug 2019 22:32:15 +0800
Subject: ASoC: max98926: remove two unused variables

sound/soc/codecs/max98926.c:28:26: warning:
 max98926_dai_txt defined but not used [-Wunused-const-variable=]
sound/soc/codecs/max98926.c:23:27: warning:
 max98926_boost_current_txt defined but not used [-Wunused-const-variable=]

They are never used, so can be removd.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190808143215.65904-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98926.c | 9 ---------
 1 file changed, 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c
index 818c0301fb29..c4dfa8ab1d49 100644
--- a/sound/soc/codecs/max98926.c
+++ b/sound/soc/codecs/max98926.c
@@ -20,15 +20,6 @@ static const char * const max98926_boost_voltage_txt[] = {
 	"6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V"
 };
 
-static const char * const max98926_boost_current_txt[] = {
-	"0.6", "0.8", "1.0", "1.2", "1.4", "1.6", "1.8", "2.0",
-	"2.2", "2.4", "2.6", "2.8", "3.2", "3.6", "4.0", "4.4"
-};
-
-static const char *const max98926_dai_txt[] = {
-	"Left", "Right", "LeftRight", "LeftRightDiv2",
-};
-
 static const char *const max98926_pdm_ch_text[] = {
 	"Current", "Voltage",
 };
-- 
cgit v1.2.3-59-g8ed1b


From c86102a333f77dcd0f7ef20ba836c6f13f1a077a Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 8 Aug 2019 22:35:07 +0800
Subject: ASoC: max9850: remove unused variable 'max9850_reg'

sound/soc/codecs/max9850.c:31:33: warning:
 max9850_reg defined but not used [-Wunused-const-variable=]

It is not used since commit 068416620c0d ("ASoC:
max9850: Convert to direct regmap API usage")

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190808143507.66788-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max9850.c | 13 -------------
 1 file changed, 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index f50ee8f5fe93..6f43748f9239 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -27,19 +27,6 @@ struct max9850_priv {
 	unsigned int sysclk;
 };
 
-/* max9850 register cache */
-static const struct reg_default max9850_reg[] = {
-	{  2, 0x0c },
-	{  3, 0x00 },
-	{  4, 0x00 },
-	{  5, 0x00 },
-	{  6, 0x00 },
-	{  7, 0x00 },
-	{  8, 0x00 },
-	{  9, 0x00 },
-	{ 10, 0x00 },
-};
-
 /* these registers are not used at the moment but provided for the sake of
  * completeness */
 static bool max9850_volatile_register(struct device *dev, unsigned int reg)
-- 
cgit v1.2.3-59-g8ed1b


From b812cd5864548bd9718879896fdd2822d3cb3d76 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:53:52 +0900
Subject: ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use
 snd_soc_dai_link_component for aux_dev

We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgqc6wa8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 4c816c86844b..53f54078f78c 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -343,7 +343,7 @@ static int
 mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *cpnt);
 
 static struct snd_soc_aux_dev mt8183_mt6358_ts3a227_max98357_headset_dev = {
-	.name = "Headset Chip",
+	.dlc = COMP_EMPTY(),
 	.init = mt8183_mt6358_ts3a227_max98357_headset_init,
 };
 
@@ -399,10 +399,10 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev)
 		dai_link->platforms->of_node = platform_node;
 	}
 
-	mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node =
+	mt8183_mt6358_ts3a227_max98357_headset_dev.dlc.of_node =
 		of_parse_phandle(pdev->dev.of_node,
 				 "mediatek,headset-codec", 0);
-	if (mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) {
+	if (mt8183_mt6358_ts3a227_max98357_headset_dev.dlc.of_node) {
 		card->aux_dev = &mt8183_mt6358_ts3a227_max98357_headset_dev;
 		card->num_aux_devs = 1;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From c2b71c71037bea7765aa6ff37824520d19108769 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 8 Aug 2019 14:54:44 +0900
Subject: ASoC: soc-core: add for_each_xxx macro for aux_dev

To be more readable code, this patch adds
new for_each_xxx() macro for aux_dev.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ftmc6w8s.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h        | 10 ++++++++++
 sound/soc/meson/axg-card.c |  7 ++++---
 sound/soc/soc-core.c       | 15 ++++++++-------
 3 files changed, 22 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 2fc56e5963f3..b1fe5ebea257 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1087,6 +1087,10 @@ struct snd_soc_card {
 	for ((i) = 0;							\
 	     ((i) < (card)->num_links) && ((link) = &(card)->dai_link[i]); \
 	     (i)++)
+#define for_each_card_pre_auxs(card, i, aux)				\
+	for ((i) = 0;							\
+	     ((i) < (card)->num_aux_devs) && ((aux) = &(card)->aux_dev[i]); \
+	     (i)++)
 
 #define for_each_card_links(card, link)				\
 	list_for_each_entry(link, &(card)->dai_link_list, list)
@@ -1098,6 +1102,12 @@ struct snd_soc_card {
 #define for_each_card_rtds_safe(card, rtd, _rtd)	\
 	list_for_each_entry_safe(rtd, _rtd, &(card)->rtd_list, list)
 
+#define for_each_card_auxs(card, component)			\
+	list_for_each_entry(component, &card->aux_comp_list, card_aux_list)
+#define for_each_card_auxs_safe(card, component, _comp)	\
+	list_for_each_entry_safe(component, _comp,	\
+				 &card->aux_comp_list, card_aux_list)
+
 #define for_each_card_components(card, component)			\
 	list_for_each_entry(component, &(card)->component_dev_list, card_list)
 
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 6283e5025548..1f698adde506 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -111,6 +111,7 @@ static void axg_card_clean_references(struct axg_card *priv)
 	struct snd_soc_card *card = &priv->card;
 	struct snd_soc_dai_link *link;
 	struct snd_soc_dai_link_component *codec;
+	struct snd_soc_aux_dev *aux;
 	int i, j;
 
 	if (card->dai_link) {
@@ -123,8 +124,8 @@ static void axg_card_clean_references(struct axg_card *priv)
 	}
 
 	if (card->aux_dev) {
-		for (i = 0; i < card->num_aux_devs; i++)
-			of_node_put(card->aux_dev[i].dlc.of_node);
+		for_each_card_pre_auxs(card, i, aux)
+			of_node_put(aux->dlc.of_node);
 	}
 
 	kfree(card->dai_link);
@@ -157,7 +158,7 @@ static int axg_card_add_aux_devices(struct snd_soc_card *card)
 	card->aux_dev = aux;
 	card->num_aux_devs = num;
 
-	for (i = 0; i < card->num_aux_devs; i++, aux++) {
+	for_each_card_pre_auxs(card, i, aux) {
 		aux->dlc.of_node =
 			of_parse_phandle(node, "audio-aux-devs", i);
 		if (!aux->dlc.of_node)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4af382d52675..e9f44505cc3e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1533,9 +1533,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 	return ret;
 }
 
-static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
+static int soc_bind_aux_dev(struct snd_soc_card *card,
+			    struct snd_soc_aux_dev *aux_dev)
 {
-	struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
 	struct snd_soc_component *component;
 
 	/* codecs, usually analog devices */
@@ -1544,6 +1544,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
 		return -EPROBE_DEFER;
 
 	component->init = aux_dev->init;
+	/* see for_each_card_auxs */
 	list_add(&component->card_aux_list, &card->aux_comp_list);
 
 	return 0;
@@ -1556,7 +1557,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card)
 	int ret;
 
 	for_each_comp_order(order) {
-		list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) {
+		for_each_card_auxs(card, comp) {
 			if (comp->driver->probe_order == order) {
 				ret = soc_probe_component(card,	comp);
 				if (ret < 0) {
@@ -1578,8 +1579,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 	int order;
 
 	for_each_comp_order(order) {
-		list_for_each_entry_safe(comp, _comp,
-			&card->aux_comp_list, card_aux_list) {
+		for_each_card_auxs_safe(card, comp, _comp) {
 
 			if (comp->driver->remove_order == order) {
 				soc_remove_component(comp);
@@ -1913,6 +1913,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai_link *dai_link;
+	struct snd_soc_aux_dev *aux;
 	int ret, i, order;
 
 	mutex_lock(&client_mutex);
@@ -1943,8 +1944,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* bind aux_devs too */
-	for (i = 0; i < card->num_aux_devs; i++) {
-		ret = soc_bind_aux_dev(card, i);
+	for_each_card_pre_auxs(card, i, aux) {
+		ret = soc_bind_aux_dev(card, aux);
 		if (ret != 0)
 			goto probe_end;
 	}
-- 
cgit v1.2.3-59-g8ed1b


From 8afd1a99d713ccc7cf1bd509b205c3bb60cf0b90 Mon Sep 17 00:00:00 2001
From: Keyon Jie <yang.jie@linux.intel.com>
Date: Fri, 9 Aug 2019 18:22:34 -0500
Subject: ASoC: Intel: sof-rt5682: add dmic dapm widget to support dmic PCM

We need add DAPM MIC endpoint widget "SoC DMIC" and route, to enable
DMIC PCM DAPM support.

Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190809232236.21182-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/sof_rt5682.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 64db00353e18..7285474f7d65 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -309,6 +309,7 @@ static const struct snd_soc_dapm_widget sof_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_SPK("Spk", NULL),
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
 };
 
 static const struct snd_soc_dapm_route sof_map[] = {
@@ -319,6 +320,9 @@ static const struct snd_soc_dapm_route sof_map[] = {
 	/* other jacks */
 	{ "IN1P", NULL, "Headset Mic" },
 
+	/* digital mics */
+	{"DMic", NULL, "SoC DMIC"},
+
 };
 
 static const struct snd_soc_dapm_route speaker_map[] = {
-- 
cgit v1.2.3-59-g8ed1b


From 8b209cdb696f49f7f5d6c1922bed435e54631ddd Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Fri, 9 Aug 2019 22:27:36 +0200
Subject: ASoC: remove w90x900/nuc900 platform drivers

The ARM w90x900 platform is getting removed, so this driver is obsolete.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20190809202749.742267-9-arnd@arndb.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/Kconfig               |   1 -
 sound/soc/Makefile              |   1 -
 sound/soc/nuc900/Kconfig        |  29 ---
 sound/soc/nuc900/Makefile       |  12 --
 sound/soc/nuc900/nuc900-ac97.c  | 391 ----------------------------------------
 sound/soc/nuc900/nuc900-audio.c |  73 --------
 sound/soc/nuc900/nuc900-audio.h | 108 -----------
 sound/soc/nuc900/nuc900-pcm.c   | 321 ---------------------------------
 8 files changed, 936 deletions(-)
 delete mode 100644 sound/soc/nuc900/Kconfig
 delete mode 100644 sound/soc/nuc900/Makefile
 delete mode 100644 sound/soc/nuc900/nuc900-ac97.c
 delete mode 100644 sound/soc/nuc900/nuc900-audio.c
 delete mode 100644 sound/soc/nuc900/nuc900-audio.h
 delete mode 100644 sound/soc/nuc900/nuc900-pcm.c

(limited to 'sound')

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index dc86e4073001..bdc305cece6e 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -51,7 +51,6 @@ source "sound/soc/dwc/Kconfig"
 source "sound/soc/fsl/Kconfig"
 source "sound/soc/hisilicon/Kconfig"
 source "sound/soc/jz4740/Kconfig"
-source "sound/soc/nuc900/Kconfig"
 source "sound/soc/kirkwood/Kconfig"
 source "sound/soc/img/Kconfig"
 source "sound/soc/intel/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 250a0dea9294..861a21b79484 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -39,7 +39,6 @@ obj-$(CONFIG_SND_SOC)	+= intel/
 obj-$(CONFIG_SND_SOC)	+= mediatek/
 obj-$(CONFIG_SND_SOC)	+= meson/
 obj-$(CONFIG_SND_SOC)	+= mxs/
-obj-$(CONFIG_SND_SOC)	+= nuc900/
 obj-$(CONFIG_SND_SOC)	+= kirkwood/
 obj-$(CONFIG_SND_SOC)	+= pxa/
 obj-$(CONFIG_SND_SOC)	+= qcom/
diff --git a/sound/soc/nuc900/Kconfig b/sound/soc/nuc900/Kconfig
deleted file mode 100644
index e1b22fbcb159..000000000000
--- a/sound/soc/nuc900/Kconfig
+++ /dev/null
@@ -1,29 +0,0 @@
-# SPDX-License-Identifier: GPL-2.0-only
-##
-## NUC900 series AC97 API
-##
-config SND_SOC_NUC900
-	tristate "SoC Audio for NUC900 series"
-	depends on ARCH_W90X900
-	select SND_SOC_NUC900_AC97
-	help
-	  This option enables support for AC97 mode on the NUC900 SoC.
-
-config SND_SOC_NUC900_AC97
-	tristate
-	select AC97_BUS
-	select SND_AC97_CODEC
-	select SND_SOC_AC97_BUS
-
-
-##
-## Boards
-##
-config SND_SOC_NUC900EVB
-	tristate "NUC900 AC97 support for demo board"
-	depends on SND_SOC_NUC900
-	select SND_SOC_NUC900_AC97
-	select SND_SOC_AC97_CODEC
-	help
-	  Select this option to enable audio (AC97) on the
-	  NUC900 demoboard.
diff --git a/sound/soc/nuc900/Makefile b/sound/soc/nuc900/Makefile
deleted file mode 100644
index c7ba2b9549d2..000000000000
--- a/sound/soc/nuc900/Makefile
+++ /dev/null
@@ -1,12 +0,0 @@
-# SPDX-License-Identifier: GPL-2.0
-# NUC900 series audio
-snd-soc-nuc900-pcm-objs := nuc900-pcm.o
-snd-soc-nuc900-ac97-objs := nuc900-ac97.o
-
-obj-$(CONFIG_SND_SOC_NUC900) += snd-soc-nuc900-pcm.o
-obj-$(CONFIG_SND_SOC_NUC900_AC97) += snd-soc-nuc900-ac97.o
-
-# Boards
-snd-soc-nuc900-audio-objs := nuc900-audio.o
-
-obj-$(CONFIG_SND_SOC_NUC900EVB) += snd-soc-nuc900-audio.o
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
deleted file mode 100644
index 5f2e5c069377..000000000000
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ /dev/null
@@ -1,391 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * Copyright (c) 2009-2010 Nuvoton technology corporation.
- *
- * Wan ZongShun <mcuos.com@gmail.com>
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/slab.h>
-#include <linux/device.h>
-#include <linux/delay.h>
-#include <linux/mutex.h>
-#include <linux/suspend.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-#include <linux/clk.h>
-
-#include <mach/mfp.h>
-
-#include "nuc900-audio.h"
-
-static DEFINE_MUTEX(ac97_mutex);
-struct nuc900_audio *nuc900_ac97_data;
-EXPORT_SYMBOL_GPL(nuc900_ac97_data);
-
-static int nuc900_checkready(void)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-
-	if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY))
-		return -EPERM;
-
-	return 0;
-}
-
-/* AC97 controller reads codec register */
-static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97,
-					unsigned short reg)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-	unsigned long timeout = 0x10000, val;
-
-	mutex_lock(&ac97_mutex);
-
-	val = nuc900_checkready();
-	if (val) {
-		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
-		goto out;
-	}
-
-	/* set the R_WB bit and write register index */
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, R_WB | reg);
-
-	/* set the valid frame bit and valid slots */
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
-	val |= (VALID_FRAME | SLOT1_VALID);
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val);
-
-	udelay(100);
-
-	/* polling the AC_R_FINISH */
-	while (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_R_FINISH)
-								&& --timeout)
-		mdelay(1);
-
-	if (!timeout) {
-		dev_err(nuc900_audio->dev, "AC97 read register time out !\n");
-		val = -EPERM;
-		goto out;
-	}
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ;
-	val &= ~SLOT1_VALID;
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val);
-
-	if (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS1) >> 2 != reg) {
-		dev_err(nuc900_audio->dev,
-				"R_INDEX of REG_ACTL_ACIS1 not match!\n");
-	}
-
-	udelay(100);
-	val = (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS2) & 0xFFFF);
-
-out:
-	mutex_unlock(&ac97_mutex);
-	return val;
-}
-
-/* AC97 controller writes to codec register */
-static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
-				unsigned short val)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-	unsigned long tmp, timeout = 0x10000;
-
-	mutex_lock(&ac97_mutex);
-
-	tmp = nuc900_checkready();
-	if (tmp)
-		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
-
-	/* clear the R_WB bit and write register index */
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, reg);
-
-	/* write register value */
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS2, val);
-
-	/* set the valid frame bit and valid slots */
-	tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
-	tmp |= SLOT1_VALID | SLOT2_VALID | VALID_FRAME;
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
-
-	udelay(100);
-
-	/* polling the AC_W_FINISH */
-	while ((AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_W_FINISH)
-								&& --timeout)
-		mdelay(1);
-
-	if (!timeout)
-		dev_err(nuc900_audio->dev, "AC97 write register time out !\n");
-
-	tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
-	tmp &= ~(SLOT1_VALID | SLOT2_VALID);
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
-
-	mutex_unlock(&ac97_mutex);
-
-}
-
-static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-	unsigned long val;
-
-	mutex_lock(&ac97_mutex);
-
-	/* warm reset AC 97 */
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
-	val |= AC_W_RES;
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
-
-	udelay(100);
-
-	val = nuc900_checkready();
-	if (val)
-		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
-
-	mutex_unlock(&ac97_mutex);
-}
-
-static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-	unsigned long val;
-
-	mutex_lock(&ac97_mutex);
-
-	/* reset Audio Controller */
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
-	val |= ACTL_RESET_BIT;
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
-	val &= (~ACTL_RESET_BIT);
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-
-	/* reset AC-link interface */
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
-	val |= AC_RESET;
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
-	val &= ~AC_RESET;
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-
-	/* cold reset AC 97 */
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
-	val |= AC_C_RES;
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
-	val &= (~AC_C_RES);
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
-
-	udelay(100);
-
-	mutex_unlock(&ac97_mutex);
-
-}
-
-/* AC97 controller operations */
-static struct snd_ac97_bus_ops nuc900_ac97_ops = {
-	.read		= nuc900_ac97_read,
-	.write		= nuc900_ac97_write,
-	.reset		= nuc900_ac97_cold_reset,
-	.warm_reset	= nuc900_ac97_warm_reset,
-};
-
-static int nuc900_ac97_trigger(struct snd_pcm_substream *substream,
-				int cmd, struct snd_soc_dai *dai)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-	int ret;
-	unsigned long val, tmp;
-
-	ret = 0;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-		val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
-			tmp |= (SLOT3_VALID | SLOT4_VALID | VALID_FRAME);
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
-
-			tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR);
-			tmp |= (P_DMA_END_IRQ | P_DMA_MIDDLE_IRQ);
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, tmp);
-			val |= AC_PLAY;
-		} else {
-			tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR);
-			tmp |= (R_DMA_END_IRQ | R_DMA_MIDDLE_IRQ);
-
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, tmp);
-			val |= AC_RECORD;
-		}
-
-		AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-		val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
-			tmp &= ~(SLOT3_VALID | SLOT4_VALID);
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
-
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, RESET_PRSR);
-			val &= ~AC_PLAY;
-		} else {
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, RESET_PRSR);
-			val &= ~AC_RECORD;
-		}
-
-		AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-
-		break;
-	default:
-		ret = -EINVAL;
-	}
-
-	return ret;
-}
-
-static int nuc900_ac97_probe(struct snd_soc_dai *dai)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-	unsigned long val;
-
-	mutex_lock(&ac97_mutex);
-
-	/* enable unit clock */
-	clk_enable(nuc900_audio->clk);
-
-	/* enable audio controller and AC-link interface */
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
-	val |= (IIS_AC_PIN_SEL | ACLINK_EN);
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
-
-	mutex_unlock(&ac97_mutex);
-
-	return 0;
-}
-
-static int nuc900_ac97_remove(struct snd_soc_dai *dai)
-{
-	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
-
-	clk_disable(nuc900_audio->clk);
-	return 0;
-}
-
-static const struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
-	.trigger	= nuc900_ac97_trigger,
-};
-
-static struct snd_soc_dai_driver nuc900_ac97_dai = {
-	.probe			= nuc900_ac97_probe,
-	.remove			= nuc900_ac97_remove,
-	.bus_control		= true,
-	.playback = {
-		.rates		= SNDRV_PCM_RATE_8000_48000,
-		.formats	= SNDRV_PCM_FMTBIT_S16_LE,
-		.channels_min	= 1,
-		.channels_max	= 2,
-	},
-	.capture = {
-		.rates		= SNDRV_PCM_RATE_8000_48000,
-		.formats	= SNDRV_PCM_FMTBIT_S16_LE,
-		.channels_min	= 1,
-		.channels_max	= 2,
-	},
-	.ops = &nuc900_ac97_dai_ops,
-};
-
-static const struct snd_soc_component_driver nuc900_ac97_component = {
-	.name		= "nuc900-ac97",
-};
-
-static int nuc900_ac97_drvprobe(struct platform_device *pdev)
-{
-	struct nuc900_audio *nuc900_audio;
-	int ret;
-
-	if (nuc900_ac97_data)
-		return -EBUSY;
-
-	nuc900_audio = devm_kzalloc(&pdev->dev, sizeof(struct nuc900_audio),
-				    GFP_KERNEL);
-	if (!nuc900_audio)
-		return -ENOMEM;
-
-	spin_lock_init(&nuc900_audio->lock);
-
-	nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev,
-						   nuc900_audio->res);
-	if (IS_ERR(nuc900_audio->mmio))
-		return PTR_ERR(nuc900_audio->mmio);
-
-	nuc900_audio->clk = devm_clk_get(&pdev->dev, NULL);
-	if (IS_ERR(nuc900_audio->clk)) {
-		ret = PTR_ERR(nuc900_audio->clk);
-		goto out;
-	}
-
-	ret = platform_get_irq(pdev, 0);
-	if (ret < 0)
-		goto out;
-	nuc900_audio->irq_num = ret;
-
-	nuc900_ac97_data = nuc900_audio;
-
-	ret = snd_soc_set_ac97_ops(&nuc900_ac97_ops);
-	if (ret)
-		goto out;
-
-	ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
-					 &nuc900_ac97_dai, 1);
-	if (ret)
-		goto out;
-
-	/* enbale ac97 multifunction pin */
-	mfp_set_groupg(nuc900_audio->dev, NULL);
-
-	return 0;
-
-out:
-	snd_soc_set_ac97_ops(NULL);
-	return ret;
-}
-
-static int nuc900_ac97_drvremove(struct platform_device *pdev)
-{
-	nuc900_ac97_data = NULL;
-	snd_soc_set_ac97_ops(NULL);
-
-	return 0;
-}
-
-static struct platform_driver nuc900_ac97_driver = {
-	.driver	= {
-		.name	= "nuc900-ac97",
-	},
-	.probe		= nuc900_ac97_drvprobe,
-	.remove		= nuc900_ac97_drvremove,
-};
-
-module_platform_driver(nuc900_ac97_driver);
-
-MODULE_AUTHOR("Wan ZongShun <mcuos.com@gmail.com>");
-MODULE_DESCRIPTION("NUC900 AC97 SoC driver!");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:nuc900-ac97");
diff --git a/sound/soc/nuc900/nuc900-audio.c b/sound/soc/nuc900/nuc900-audio.c
deleted file mode 100644
index 19146690d514..000000000000
--- a/sound/soc/nuc900/nuc900-audio.c
+++ /dev/null
@@ -1,73 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * Copyright (c) 2010 Nuvoton technology corporation.
- *
- * Wan ZongShun <mcuos.com@gmail.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include "nuc900-audio.h"
-
-SND_SOC_DAILINK_DEFS(ac97,
-	DAILINK_COMP_ARRAY(COMP_CPU("nuc900-ac97")),
-	DAILINK_COMP_ARRAY(COMP_CODEC("ac97-codec", "ac97-hifi")),
-	DAILINK_COMP_ARRAY(COMP_PLATFORM("nuc900-pcm-audio")));
-
-static struct snd_soc_dai_link nuc900evb_ac97_dai = {
-	.name		= "AC97",
-	.stream_name	= "AC97 HiFi",
-	SND_SOC_DAILINK_REG(ac97),
-};
-
-static struct snd_soc_card nuc900evb_audio_machine = {
-	.name		= "NUC900EVB_AC97",
-	.owner		= THIS_MODULE,
-	.dai_link	= &nuc900evb_ac97_dai,
-	.num_links	= 1,
-};
-
-static struct platform_device *nuc900evb_asoc_dev;
-
-static int __init nuc900evb_audio_init(void)
-{
-	int ret;
-
-	ret = -ENOMEM;
-	nuc900evb_asoc_dev = platform_device_alloc("soc-audio", -1);
-	if (!nuc900evb_asoc_dev)
-		goto out;
-
-	/* nuc900 board audio device */
-	platform_set_drvdata(nuc900evb_asoc_dev, &nuc900evb_audio_machine);
-
-	ret = platform_device_add(nuc900evb_asoc_dev);
-
-	if (ret) {
-		platform_device_put(nuc900evb_asoc_dev);
-		nuc900evb_asoc_dev = NULL;
-	}
-
-out:
-	return ret;
-}
-
-static void __exit nuc900evb_audio_exit(void)
-{
-	platform_device_unregister(nuc900evb_asoc_dev);
-}
-
-module_init(nuc900evb_audio_init);
-module_exit(nuc900evb_audio_exit);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("NUC900 Series ASoC audio support");
-MODULE_AUTHOR("Wan ZongShun");
diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h
deleted file mode 100644
index 90ffa7bbce01..000000000000
--- a/sound/soc/nuc900/nuc900-audio.h
+++ /dev/null
@@ -1,108 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * Copyright (c) 2010 Nuvoton technology corporation.
- *
- * Wan ZongShun <mcuos.com@gmail.com>
- */
-
-#ifndef _NUC900_AUDIO_H
-#define _NUC900_AUDIO_H
-
-#include <linux/io.h>
-
-/* Audio Control Registers */
-#define ACTL_CON		0x00
-#define ACTL_RESET		0x04
-#define ACTL_RDSTB		0x08
-#define ACTL_RDST_LENGTH	0x0C
-#define ACTL_RDSTC		0x10
-#define ACTL_RSR		0x14
-#define ACTL_PDSTB		0x18
-#define ACTL_PDST_LENGTH	0x1C
-#define ACTL_PDSTC		0x20
-#define ACTL_PSR		0x24
-#define ACTL_IISCON		0x28
-#define ACTL_ACCON		0x2C
-#define ACTL_ACOS0		0x30
-#define ACTL_ACOS1		0x34
-#define ACTL_ACOS2		0x38
-#define ACTL_ACIS0		0x3C
-#define ACTL_ACIS1		0x40
-#define ACTL_ACIS2		0x44
-#define ACTL_COUNTER		0x48
-
-/* bit definition of REG_ACTL_CON register */
-#define R_DMA_IRQ		0x1000
-#define T_DMA_IRQ		0x0800
-#define IIS_AC_PIN_SEL		0x0100
-#define FIFO_TH			0x0080
-#define ADC_EN			0x0010
-#define M80_EN			0x0008
-#define ACLINK_EN		0x0004
-#define IIS_EN			0x0002
-
-/* bit definition of REG_ACTL_RESET register */
-#define W5691_PLAY		0x20000
-#define ACTL_RESET_BIT		0x10000
-#define RECORD_RIGHT_CHNNEL	0x08000
-#define RECORD_LEFT_CHNNEL	0x04000
-#define PLAY_RIGHT_CHNNEL	0x02000
-#define PLAY_LEFT_CHNNEL	0x01000
-#define DAC_PLAY		0x00800
-#define ADC_RECORD		0x00400
-#define M80_PLAY		0x00200
-#define AC_RECORD		0x00100
-#define AC_PLAY			0x00080
-#define IIS_RECORD		0x00040
-#define IIS_PLAY		0x00020
-#define DAC_RESET		0x00010
-#define ADC_RESET		0x00008
-#define M80_RESET		0x00004
-#define AC_RESET		0x00002
-#define IIS_RESET		0x00001
-
-/* bit definition of REG_ACTL_ACCON register */
-#define AC_BCLK_PU_EN		0x20
-#define AC_R_FINISH		0x10
-#define AC_W_FINISH		0x08
-#define AC_W_RES		0x04
-#define AC_C_RES		0x02
-
-/* bit definition of ACTL_RSR register */
-#define R_FIFO_EMPTY		0x04
-#define R_DMA_END_IRQ		0x02
-#define R_DMA_MIDDLE_IRQ	0x01
-
-/* bit definition of ACTL_PSR register */
-#define P_FIFO_EMPTY		0x04
-#define P_DMA_END_IRQ		0x02
-#define P_DMA_MIDDLE_IRQ	0x01
-
-/* bit definition of ACTL_ACOS0 register */
-#define SLOT1_VALID		0x01
-#define SLOT2_VALID		0x02
-#define SLOT3_VALID		0x04
-#define SLOT4_VALID		0x08
-#define VALID_FRAME		0x10
-
-/* bit definition of ACTL_ACOS1 register */
-#define R_WB			0x80
-
-#define CODEC_READY		0x10
-#define RESET_PRSR		0x00
-#define AUDIO_WRITE(addr, val)	__raw_writel(val, addr)
-#define AUDIO_READ(addr)	__raw_readl(addr)
-
-struct nuc900_audio {
-	void __iomem *mmio;
-	spinlock_t lock;
-	unsigned long irq_num;
-	struct resource *res;
-	struct clk *clk;
-	struct device *dev;
-
-};
-
-extern struct nuc900_audio *nuc900_ac97_data;
-
-#endif /*end _NUC900_AUDIO_H */
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
deleted file mode 100644
index 4442a26e9502..000000000000
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ /dev/null
@@ -1,321 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * Copyright (c) 2010 Nuvoton technology corporation.
- *
- * Wan ZongShun <mcuos.com@gmail.com>
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/io.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/hardware.h>
-
-#include "nuc900-audio.h"
-
-static const struct snd_pcm_hardware nuc900_pcm_hardware = {
-	.info			= SNDRV_PCM_INFO_INTERLEAVED |
-					SNDRV_PCM_INFO_BLOCK_TRANSFER |
-					SNDRV_PCM_INFO_MMAP |
-					SNDRV_PCM_INFO_MMAP_VALID |
-					SNDRV_PCM_INFO_PAUSE |
-					SNDRV_PCM_INFO_RESUME,
-	.buffer_bytes_max	= 4*1024,
-	.period_bytes_min	= 1*1024,
-	.period_bytes_max	= 4*1024,
-	.periods_min		= 1,
-	.periods_max		= 1024,
-};
-
-static int nuc900_dma_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
-}
-
-static void nuc900_update_dma_register(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct nuc900_audio *nuc900_audio = runtime->private_data;
-	void __iomem *mmio_addr, *mmio_len;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		mmio_addr = nuc900_audio->mmio + ACTL_PDSTB;
-		mmio_len = nuc900_audio->mmio + ACTL_PDST_LENGTH;
-	} else {
-		mmio_addr = nuc900_audio->mmio + ACTL_RDSTB;
-		mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH;
-	}
-
-	AUDIO_WRITE(mmio_addr, runtime->dma_addr);
-	AUDIO_WRITE(mmio_len, runtime->dma_bytes);
-}
-
-static void nuc900_dma_start(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct nuc900_audio *nuc900_audio = runtime->private_data;
-	unsigned long val;
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
-	val |= (T_DMA_IRQ | R_DMA_IRQ);
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
-}
-
-static void nuc900_dma_stop(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct nuc900_audio *nuc900_audio = runtime->private_data;
-	unsigned long val;
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
-	val &= ~(T_DMA_IRQ | R_DMA_IRQ);
-	AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
-}
-
-static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id)
-{
-	struct snd_pcm_substream *substream = dev_id;
-	struct nuc900_audio *nuc900_audio = substream->runtime->private_data;
-	unsigned long val;
-
-	spin_lock(&nuc900_audio->lock);
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
-
-	if (val & R_DMA_IRQ) {
-		AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | R_DMA_IRQ);
-
-		val = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR);
-
-		if (val & R_DMA_MIDDLE_IRQ) {
-			val |= R_DMA_MIDDLE_IRQ;
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val);
-		}
-
-		if (val & R_DMA_END_IRQ) {
-			val |= R_DMA_END_IRQ;
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val);
-		}
-	} else if (val & T_DMA_IRQ) {
-		AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | T_DMA_IRQ);
-
-		val = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR);
-
-		if (val & P_DMA_MIDDLE_IRQ) {
-			val |= P_DMA_MIDDLE_IRQ;
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val);
-		}
-
-		if (val & P_DMA_END_IRQ) {
-			val |= P_DMA_END_IRQ;
-			AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val);
-		}
-	} else {
-		dev_err(nuc900_audio->dev, "Wrong DMA interrupt status!\n");
-		spin_unlock(&nuc900_audio->lock);
-		return IRQ_HANDLED;
-	}
-
-	spin_unlock(&nuc900_audio->lock);
-
-	snd_pcm_period_elapsed(substream);
-
-	return IRQ_HANDLED;
-}
-
-static int nuc900_dma_hw_free(struct snd_pcm_substream *substream)
-{
-	snd_pcm_lib_free_pages(substream);
-	return 0;
-}
-
-static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct nuc900_audio *nuc900_audio = runtime->private_data;
-	unsigned long flags, val;
-	int ret = 0;
-
-	spin_lock_irqsave(&nuc900_audio->lock, flags);
-
-	nuc900_update_dma_register(substream);
-
-	val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
-
-	switch (runtime->channels) {
-	case 1:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			val &= ~(PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL);
-			val |= PLAY_RIGHT_CHNNEL;
-		} else {
-			val &= ~(RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL);
-			val |= RECORD_RIGHT_CHNNEL;
-		}
-		AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-		break;
-	case 2:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			val |= (PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL);
-		else
-			val |= (RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL);
-		AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
-		break;
-	default:
-		ret = -EINVAL;
-	}
-	spin_unlock_irqrestore(&nuc900_audio->lock, flags);
-	return ret;
-}
-
-static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	int ret = 0;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-		nuc900_dma_start(substream);
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-		nuc900_dma_stop(substream);
-		break;
-
-	default:
-		ret = -EINVAL;
-		break;
-	}
-
-	return ret;
-}
-
-static int nuc900_dma_getposition(struct snd_pcm_substream *substream,
-					dma_addr_t *src, dma_addr_t *dst)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct nuc900_audio *nuc900_audio = runtime->private_data;
-
-	if (src != NULL)
-		*src = AUDIO_READ(nuc900_audio->mmio + ACTL_PDSTC);
-
-	if (dst != NULL)
-		*dst = AUDIO_READ(nuc900_audio->mmio + ACTL_RDSTC);
-
-	return 0;
-}
-
-static snd_pcm_uframes_t nuc900_dma_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	dma_addr_t src, dst;
-	unsigned long res;
-
-	nuc900_dma_getposition(substream, &src, &dst);
-
-	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-		res = dst - runtime->dma_addr;
-	else
-		res = src - runtime->dma_addr;
-
-	return bytes_to_frames(substream->runtime, res);
-}
-
-static int nuc900_dma_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct nuc900_audio *nuc900_audio;
-
-	snd_soc_set_runtime_hwparams(substream, &nuc900_pcm_hardware);
-
-	nuc900_audio = nuc900_ac97_data;
-
-	if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt,
-			0, "nuc900-dma", substream))
-		return -EBUSY;
-
-	runtime->private_data = nuc900_audio;
-
-	return 0;
-}
-
-static int nuc900_dma_close(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct nuc900_audio *nuc900_audio = runtime->private_data;
-
-	free_irq(nuc900_audio->irq_num, substream);
-
-	return 0;
-}
-
-static int nuc900_dma_mmap(struct snd_pcm_substream *substream,
-	struct vm_area_struct *vma)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	return dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area,
-			   runtime->dma_addr, runtime->dma_bytes);
-}
-
-static const struct snd_pcm_ops nuc900_dma_ops = {
-	.open		= nuc900_dma_open,
-	.close		= nuc900_dma_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= nuc900_dma_hw_params,
-	.hw_free	= nuc900_dma_hw_free,
-	.prepare	= nuc900_dma_prepare,
-	.trigger	= nuc900_dma_trigger,
-	.pointer	= nuc900_dma_pointer,
-	.mmap		= nuc900_dma_mmap,
-};
-
-static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_card *card = rtd->card->snd_card;
-	struct snd_pcm *pcm = rtd->pcm;
-	int ret;
-
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		return ret;
-
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
-		card->dev, 4 * 1024, (4 * 1024) - 1);
-
-	return 0;
-}
-
-static const struct snd_soc_component_driver nuc900_soc_component = {
-	.ops		= &nuc900_dma_ops,
-	.pcm_new	= nuc900_dma_new,
-};
-
-static int nuc900_soc_platform_probe(struct platform_device *pdev)
-{
-	return devm_snd_soc_register_component(&pdev->dev, &nuc900_soc_component,
-					       NULL, 0);
-}
-
-static struct platform_driver nuc900_pcm_driver = {
-	.driver = {
-			.name = "nuc900-pcm-audio",
-	},
-
-	.probe = nuc900_soc_platform_probe,
-};
-
-module_platform_driver(nuc900_pcm_driver);
-
-MODULE_AUTHOR("Wan ZongShun, <mcuos.com@gmail.com>");
-MODULE_DESCRIPTION("nuc900 Audio DMA module");
-MODULE_LICENSE("GPL");
-- 
cgit v1.2.3-59-g8ed1b


From ae032156ec4142b7a1d644eba652806785de1777 Mon Sep 17 00:00:00 2001
From: Curtis Malainey <cujomalainey@chromium.org>
Date: Fri, 9 Aug 2019 14:59:51 -0700
Subject: ASoC: rt5677: Revert remove superfluous set

Commit 4dc057a786dc23 ("ASoC: rt5677: remove superfluous set") was an
attempted code cleanup but was incorrectly tested before sent and
actually breaks the interrupt since it never resets the value on each
loop now. The breakage is most testable when hotwording code is added
and also uses the interrupt.

Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20190809215952.155660-1-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index deaed5132dc9..c779dc3474f9 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -5098,6 +5098,7 @@ static irqreturn_t rt5677_irq(int unused, void *data)
 			goto exit;
 		}
 
+		irq_fired = false;
 		for (i = 0; i < RT5677_IRQ_NUM; i++) {
 			if (reg_irq & rt5677_irq_descs[i].status_mask) {
 				irq_fired = true;
-- 
cgit v1.2.3-59-g8ed1b


From 9c2806c4941641a6c75736f8c4303c89d2013cc4 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Fri, 9 Aug 2019 18:27:46 +0800
Subject: ASoC: fsl_esai: Add compatible string for imx6ull

Add compatible string for imx6ull, from imx6ull platform,
the issue of channel swap after xrun is fixed in hardware.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1565346467-5769-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_esai.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 5832144beb9f..a78e4ab478df 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -1049,6 +1049,7 @@ static int fsl_esai_remove(struct platform_device *pdev)
 static const struct of_device_id fsl_esai_dt_ids[] = {
 	{ .compatible = "fsl,imx35-esai", },
 	{ .compatible = "fsl,vf610-esai", },
+	{ .compatible = "fsl,imx6ull-esai", },
 	{}
 };
 MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids);
-- 
cgit v1.2.3-59-g8ed1b


From cac974a51ebb955f3be260a9633836e1413c8672 Mon Sep 17 00:00:00 2001
From: Jaska Uimonen <jaska.uimonen@intel.com>
Date: Fri, 9 Aug 2019 18:17:14 -0500
Subject: ASoC: SOF: topology: use set_get_data in process load

Currently when loading sof process components there's a check if binary
control data is associated with it. If found the data is extracted to be
part of component loading and initialization. If binary data exceeds the
ipc max size, loading fails with error as large message support is only
implemented in set_get_data method. So make the process loading use
set_get_data to enable large parameters in component initialization.

Also refactor the process component loading function as it digs out 3
times almost identical information of related controls. This is
redundant, looks ugly and makes it difficult to understand the
mechanism. So make a function out of fetching the control data and use
it in process loading.

Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Jaska Uimonen <jaska.uimonen@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190809231714.20874-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/topology.c | 201 ++++++++++++++++++++++++++++++-----------------
 1 file changed, 127 insertions(+), 74 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 12b7d900b9c2..9cffea142395 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -42,6 +42,13 @@
 /* size of tplg abi in byte */
 #define SOF_TPLG_ABI_SIZE 3
 
+struct sof_widget_data {
+	int ctrl_type;
+	int ipc_cmd;
+	struct sof_abi_hdr *pdata;
+	struct snd_sof_control *control;
+};
+
 /* send pcm params ipc */
 static int ipc_pcm_params(struct snd_sof_widget *swidget, int dir)
 {
@@ -1742,51 +1749,32 @@ err:
 	return ret;
 }
 
-static int sof_process_load(struct snd_soc_component *scomp, int index,
-			    struct snd_sof_widget *swidget,
-			    struct snd_soc_tplg_dapm_widget *tw,
-			    struct sof_ipc_comp_reply *r,
-			    int type)
+static size_t sof_get_control_data(struct snd_sof_dev *sdev,
+				   struct snd_soc_dapm_widget *widget,
+				   struct sof_widget_data *wdata)
 {
-	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
-	struct snd_soc_tplg_private *private = &tw->priv;
-	struct snd_soc_dapm_widget *widget = swidget->widget;
 	const struct snd_kcontrol_new *kc;
-	struct soc_bytes_ext *sbe;
 	struct soc_mixer_control *sm;
+	struct soc_bytes_ext *sbe;
 	struct soc_enum *se;
-	struct snd_sof_control *scontrol = NULL;
-	struct sof_abi_hdr *pdata = NULL;
-	struct sof_ipc_comp_process *process;
-	size_t ipc_size, ipc_data_size = 0;
-	int ret, i, offset = 0;
-
-	if (type == SOF_COMP_NONE) {
-		dev_err(sdev->dev, "error: invalid process comp type %d\n",
-			type);
-		return -EINVAL;
-	}
+	size_t size = 0;
+	int i;
 
-	/*
-	 * get possible component controls - get size of all pdata,
-	 * then memcpy with headers
-	 */
 	for (i = 0; i < widget->num_kcontrols; i++) {
-
 		kc = &widget->kcontrol_news[i];
 
 		switch (widget->dobj.widget.kcontrol_type) {
 		case SND_SOC_TPLG_TYPE_MIXER:
 			sm = (struct soc_mixer_control *)kc->private_value;
-			scontrol = sm->dobj.private;
+			wdata[i].control = sm->dobj.private;
 			break;
 		case SND_SOC_TPLG_TYPE_BYTES:
 			sbe = (struct soc_bytes_ext *)kc->private_value;
-			scontrol = sbe->dobj.private;
+			wdata[i].control = sbe->dobj.private;
 			break;
 		case SND_SOC_TPLG_TYPE_ENUM:
 			se = (struct soc_enum *)kc->private_value;
-			scontrol = se->dobj.private;
+			wdata[i].control = se->dobj.private;
 			break;
 		default:
 			dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n",
@@ -1795,31 +1783,98 @@ static int sof_process_load(struct snd_soc_component *scomp, int index,
 			return -EINVAL;
 		}
 
-		if (!scontrol) {
+		if (!wdata[i].control) {
 			dev_err(sdev->dev, "error: no scontrol for widget %s\n",
 				widget->name);
 			return -EINVAL;
 		}
 
-		/* don't include if no private data */
-		pdata = scontrol->control_data->data;
-		if (!pdata)
-			continue;
+		wdata[i].pdata = wdata[i].control->control_data->data;
+		if (!wdata[i].pdata)
+			return -EINVAL;
 
 		/* make sure data is valid - data can be updated at runtime */
-		if (pdata->magic != SOF_ABI_MAGIC)
-			continue;
+		if (wdata[i].pdata->magic != SOF_ABI_MAGIC)
+			return -EINVAL;
+
+		size += wdata[i].pdata->size;
+
+		/* get data type */
+		switch (wdata[i].control->cmd) {
+		case SOF_CTRL_CMD_VOLUME:
+		case SOF_CTRL_CMD_ENUM:
+		case SOF_CTRL_CMD_SWITCH:
+			wdata[i].ipc_cmd = SOF_IPC_COMP_SET_VALUE;
+			wdata[i].ctrl_type = SOF_CTRL_TYPE_VALUE_CHAN_SET;
+			break;
+		case SOF_CTRL_CMD_BINARY:
+			wdata[i].ipc_cmd = SOF_IPC_COMP_SET_DATA;
+			wdata[i].ctrl_type = SOF_CTRL_TYPE_DATA_SET;
+			break;
+		default:
+			break;
+		}
+	}
+
+	return size;
+}
+
+static int sof_process_load(struct snd_soc_component *scomp, int index,
+			    struct snd_sof_widget *swidget,
+			    struct snd_soc_tplg_dapm_widget *tw,
+			    struct sof_ipc_comp_reply *r,
+			    int type)
+{
+	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
+	struct snd_soc_dapm_widget *widget = swidget->widget;
+	struct snd_soc_tplg_private *private = &tw->priv;
+	struct sof_ipc_comp_process *process = NULL;
+	struct sof_widget_data *wdata = NULL;
+	size_t ipc_data_size = 0;
+	size_t ipc_size;
+	int offset = 0;
+	int ret = 0;
+	int i;
+
+	if (type == SOF_COMP_NONE) {
+		dev_err(sdev->dev, "error: invalid process comp type %d\n",
+			type);
+		return -EINVAL;
+	}
+
+	/* allocate struct for widget control data sizes and types */
+	if (widget->num_kcontrols) {
+		wdata = kcalloc(widget->num_kcontrols,
+				sizeof(*wdata),
+				GFP_KERNEL);
+
+		if (!wdata)
+			return -ENOMEM;
 
-		ipc_data_size += pdata->size;
+		/* get possible component controls and get size of all pdata */
+		ipc_data_size = sof_get_control_data(sdev, widget, wdata);
+
+		if (ipc_data_size <= 0) {
+			ret = ipc_data_size;
+			goto out;
+		}
 	}
 
 	ipc_size = sizeof(struct sof_ipc_comp_process) +
 		le32_to_cpu(private->size) +
 		ipc_data_size;
 
+	/* we are exceeding max ipc size, config needs to be sent separately */
+	if (ipc_size > SOF_IPC_MSG_MAX_SIZE) {
+		ipc_size -= ipc_data_size;
+		ipc_data_size = 0;
+	}
+
 	process = kzalloc(ipc_size, GFP_KERNEL);
-	if (!process)
-		return -ENOMEM;
+	if (!process) {
+		ret = -ENOMEM;
+		goto out;
+	}
 
 	/* configure iir IPC message */
 	process->comp.hdr.size = ipc_size;
@@ -1845,40 +1900,13 @@ static int sof_process_load(struct snd_soc_component *scomp, int index,
 	 * get possible component controls - get size of all pdata,
 	 * then memcpy with headers
 	 */
-	for (i = 0; i < widget->num_kcontrols; i++) {
-		kc = &widget->kcontrol_news[i];
-
-		switch (widget->dobj.widget.kcontrol_type) {
-		case SND_SOC_TPLG_TYPE_MIXER:
-			sm = (struct soc_mixer_control *)kc->private_value;
-			scontrol = sm->dobj.private;
-			break;
-		case SND_SOC_TPLG_TYPE_BYTES:
-			sbe = (struct soc_bytes_ext *)kc->private_value;
-			scontrol = sbe->dobj.private;
-			break;
-		case SND_SOC_TPLG_TYPE_ENUM:
-			se = (struct soc_enum *)kc->private_value;
-			scontrol = se->dobj.private;
-			break;
-		default:
-			dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n",
-				widget->dobj.widget.kcontrol_type,
-				widget->name);
-			return -EINVAL;
+	if (ipc_data_size) {
+		for (i = 0; i < widget->num_kcontrols; i++) {
+			memcpy(&process->data + offset,
+			       wdata[i].pdata->data,
+			       wdata[i].pdata->size);
+			offset += wdata[i].pdata->size;
 		}
-
-		/* don't include if no private data */
-		pdata = scontrol->control_data->data;
-		if (!pdata)
-			continue;
-
-		/* make sure data is valid - data can be updated at runtime */
-		if (pdata->magic != SOF_ABI_MAGIC)
-			continue;
-
-		memcpy(&process->data + offset, pdata->data, pdata->size);
-		offset += pdata->size;
 	}
 
 	process->size = ipc_data_size;
@@ -1886,10 +1914,35 @@ static int sof_process_load(struct snd_soc_component *scomp, int index,
 
 	ret = sof_ipc_tx_message(sdev->ipc, process->comp.hdr.cmd, process,
 				 ipc_size, r, sizeof(*r));
-	if (ret >= 0)
-		return ret;
+
+	if (ret < 0) {
+		dev_err(sdev->dev, "error: create process failed\n");
+		goto err;
+	}
+
+	/* we sent the data in single message so return */
+	if (ipc_data_size)
+		goto out;
+
+	/* send control data with large message supported method */
+	for (i = 0; i < widget->num_kcontrols; i++) {
+		wdata[i].control->readback_offset = 0;
+		ret = snd_sof_ipc_set_get_comp_data(sdev->ipc, wdata[i].control,
+						    wdata[i].ipc_cmd,
+						    wdata[i].ctrl_type,
+						    wdata[i].control->cmd,
+						    true);
+		if (ret != 0) {
+			dev_err(sdev->dev, "error: send control failed\n");
+			break;
+		}
+	}
+
 err:
-	kfree(process);
+	if (ret < 0)
+		kfree(process);
+out:
+	kfree(wdata);
 	return ret;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 872f3ac583cb00c14a4dc86b1511cf98618dd671 Mon Sep 17 00:00:00 2001
From: Szymon Mielczarek <szymonx.mielczarek@linux.intel.com>
Date: Fri, 9 Aug 2019 10:40:34 +0200
Subject: ASoC: dapm: Invalidate only paths reachable for a given stream

By resetting the cached number of endpoints for all card's widgets we may
overwrite previously cached values for other streams. The situation may
happen especially when running streams simultaneously.

Signed-off-by: Szymon Mielczarek <szymonx.mielczarek@linux.intel.com>
Link: https://lore.kernel.org/r/20190809084034.26220-1-szymonx.mielczarek@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 50 +++++++++++++++++++++++++++++++++++++-------------
 1 file changed, 37 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d09bdca63c62..10819b3e0b98 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1128,6 +1128,34 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list,
 	return 0;
 }
 
+/*
+ * Recursively reset the cached number of inputs or outputs for the specified
+ * widget and all widgets that can be reached via incoming or outcoming paths
+ * from the widget.
+ */
+static void invalidate_paths_ep(struct snd_soc_dapm_widget *widget,
+	enum snd_soc_dapm_direction dir)
+{
+	enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir);
+	struct snd_soc_dapm_path *path;
+
+	widget->endpoints[dir] = -1;
+
+	snd_soc_dapm_widget_for_each_path(widget, rdir, path) {
+		if (path->weak || path->is_supply)
+			continue;
+
+		if (path->walking)
+			return;
+
+		if (path->connect) {
+			path->walking = 1;
+			invalidate_paths_ep(path->node[dir], dir);
+			path->walking = 0;
+		}
+	}
+}
+
 /*
  * Common implementation for is_connected_output_ep() and
  * is_connected_input_ep(). The function is inlined since the combined size of
@@ -1257,21 +1285,17 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
-	/*
-	 * For is_connected_{output,input}_ep fully discover the graph we need
-	 * to reset the cached number of inputs and outputs.
-	 */
-	list_for_each_entry(w, &card->widgets, list) {
-		w->endpoints[SND_SOC_DAPM_DIR_IN] = -1;
-		w->endpoints[SND_SOC_DAPM_DIR_OUT] = -1;
-	}
-
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-		paths = is_connected_output_ep(dai->playback_widget, &widgets,
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		w = dai->playback_widget;
+		invalidate_paths_ep(w, SND_SOC_DAPM_DIR_OUT);
+		paths = is_connected_output_ep(w, &widgets,
 				custom_stop_condition);
-	else
-		paths = is_connected_input_ep(dai->capture_widget, &widgets,
+	} else {
+		w = dai->capture_widget;
+		invalidate_paths_ep(w, SND_SOC_DAPM_DIR_IN);
+		paths = is_connected_input_ep(w, &widgets,
 				custom_stop_condition);
+	}
 
 	/* Drop starting point */
 	list_del(widgets.next);
-- 
cgit v1.2.3-59-g8ed1b


From 1a3fb5d3f420d5f022768fcff091e5a9862f5877 Mon Sep 17 00:00:00 2001
From: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Date: Fri, 9 Aug 2019 18:22:33 -0500
Subject: ASoC: Intel: boards: Match Product Family instead of product

The generic machine driver of sof_rt5682
supports more platforms of same product family.
hence match the product family instead of product name.

Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190809232236.21182-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/sof_rt5682.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index daeaa396d928..c1181bc0119e 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -91,8 +91,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
 	{
 		.callback = sof_rt5682_quirk_cb,
 		.matches = {
-			DMI_MATCH(DMI_SYS_VENDOR, "Google"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "Hatch"),
+			DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Hatch"),
 		},
 		.driver_data = (void *)(SOF_RT5682_MCLK_EN |
 					SOF_RT5682_MCLK_24MHZ |
-- 
cgit v1.2.3-59-g8ed1b


From 0181d2853126db0717de6bd5b3287f1c87faf7c3 Mon Sep 17 00:00:00 2001
From: Keyon Jie <yang.jie@linux.intel.com>
Date: Fri, 9 Aug 2019 18:22:35 -0500
Subject: ASoC: Intel: skl-hda-dsp-generic: add dependency to dmic driver

The hda generic machine actually has dependency on the dmic driver,
select SND_SOC_DMIC at the machine selected to fix it.

Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190809232236.21182-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/Kconfig | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 50bf149818b5..4a529cea0190 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -387,6 +387,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC
 config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
 	tristate "SKL/KBL/BXT/APL with HDA Codecs"
 	select SND_SOC_HDAC_HDMI
+	select SND_SOC_DMIC
 	# SND_SOC_HDAC_HDA is already selected
 	help
 	  This adds support for ASoC machine driver for Intel platforms
-- 
cgit v1.2.3-59-g8ed1b


From 79631210fc413546d0ed73632ff904ded5192cc9 Mon Sep 17 00:00:00 2001
From: Keyon Jie <yang.jie@linux.intel.com>
Date: Fri, 9 Aug 2019 18:22:36 -0500
Subject: ASoC: Intel: skl-hda-dsp-generic: add dmic dapm widget and route

Adding DAPM MIC endpoint widget "SoC DMIC" and route, to enable
DMIC DAPM support with hda generic machine.

Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190809232236.21182-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 9ed68eb4f058..1778acdc367c 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -23,6 +23,7 @@ static const struct snd_soc_dapm_widget skl_hda_widgets[] = {
 	SND_SOC_DAPM_MIC("Alt Analog In", NULL),
 	SND_SOC_DAPM_SPK("Digital Out", NULL),
 	SND_SOC_DAPM_MIC("Digital In", NULL),
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
 };
 
 static const struct snd_soc_dapm_route skl_hda_map[] = {
@@ -41,6 +42,9 @@ static const struct snd_soc_dapm_route skl_hda_map[] = {
 	{ "Codec Input Pin2", NULL, "Digital In" },
 	{ "Codec Input Pin3", NULL, "Alt Analog In" },
 
+	/* digital mics */
+	{"DMic", NULL, "SoC DMIC"},
+
 	/* CODEC BE connections */
 	{ "Analog Codec Playback", NULL, "Analog CPU Playback" },
 	{ "Analog CPU Playback", NULL, "codec0_out" },
-- 
cgit v1.2.3-59-g8ed1b


From abe51c351827e0086dad079dfe02918fecdf4830 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Mon, 12 Aug 2019 12:52:25 +0300
Subject: ASoC: pcm3168a: Retain the independence of DAC and ADC side of the
 codec

The DAC and ADC path of the codec is independent, have dedicated LRCK (FS)
and BCK for DAC/ADC.

They can be configured to use different format, TDM slots and slot_width if
needed.

Move these parameters under dedicated io_params structure and manage them
independently based on the dai.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190812095226.18870-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/pcm3168a.c | 125 +++++++++++++++++++-------------------------
 1 file changed, 54 insertions(+), 71 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index e84a1509fe65..75fa8e9ee894 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -44,18 +44,25 @@ static const char *const pcm3168a_supply_names[PCM3168A_NUM_SUPPLIES] = {
 	"VCCDA2"
 };
 
+#define PCM3168A_DAI_DAC		0
+#define PCM3168A_DAI_ADC		1
+
+/* ADC/DAC side parameters */
+struct pcm3168a_io_params {
+	bool master_mode;
+	unsigned int fmt;
+	int tdm_slots;
+	u32 tdm_mask;
+	int slot_width;
+};
+
 struct pcm3168a_priv {
 	struct regulator_bulk_data supplies[PCM3168A_NUM_SUPPLIES];
 	struct regmap *regmap;
 	struct clk *scki;
-	bool adc_master_mode;
-	bool dac_master_mode;
 	unsigned long sysclk;
-	unsigned int adc_fmt;
-	unsigned int dac_fmt;
-	int tdm_slots;
-	u32 tdm_mask[2];
-	int slot_width;
+
+	struct pcm3168a_io_params io_params[2];
 };
 
 static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" };
@@ -308,8 +315,7 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai,
-			       unsigned int format, bool dac)
+static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format)
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
@@ -356,43 +362,31 @@ static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	if (dac) {
+	if (dai->id == PCM3168A_DAI_DAC) {
 		reg = PCM3168A_DAC_PWR_MST_FMT;
 		mask = PCM3168A_DAC_FMT_MASK;
 		shift = PCM3168A_DAC_FMT_SHIFT;
-		pcm3168a->dac_master_mode = master_mode;
-		pcm3168a->dac_fmt = fmt;
 	} else {
 		reg = PCM3168A_ADC_MST_FMT;
 		mask = PCM3168A_ADC_FMTAD_MASK;
 		shift = PCM3168A_ADC_FMTAD_SHIFT;
-		pcm3168a->adc_master_mode = master_mode;
-		pcm3168a->adc_fmt = fmt;
 	}
 
+	pcm3168a->io_params[dai->id].master_mode = master_mode;
+	pcm3168a->io_params[dai->id].fmt = fmt;
+
 	regmap_update_bits(pcm3168a->regmap, reg, mask, fmt << shift);
 
 	return 0;
 }
 
-static int pcm3168a_set_dai_fmt_dac(struct snd_soc_dai *dai,
-			       unsigned int format)
-{
-	return pcm3168a_set_dai_fmt(dai, format, true);
-}
-
-static int pcm3168a_set_dai_fmt_adc(struct snd_soc_dai *dai,
-			       unsigned int format)
-{
-	return pcm3168a_set_dai_fmt(dai, format, false);
-}
-
 static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 				 unsigned int rx_mask, int slots,
 				 int slot_width)
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
+	struct pcm3168a_io_params *io_params = &pcm3168a->io_params[dai->id];
 
 	if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
 		dev_err(component->dev,
@@ -408,22 +402,25 @@ static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		return -EINVAL;
 	}
 
-	if (pcm3168a->tdm_slots && pcm3168a->tdm_slots != slots) {
+	if (io_params->tdm_slots && io_params->tdm_slots != slots) {
 		dev_err(component->dev, "Not matching slots %d vs %d\n",
-			pcm3168a->tdm_slots, slots);
+			io_params->tdm_slots, slots);
 		return -EINVAL;
 	}
 
-	if (pcm3168a->slot_width && pcm3168a->slot_width != slot_width) {
+	if (io_params->slot_width && io_params->slot_width != slot_width) {
 		dev_err(component->dev, "Not matching slot_width %d vs %d\n",
-			pcm3168a->slot_width, slot_width);
+			io_params->slot_width, slot_width);
 		return -EINVAL;
 	}
 
-	pcm3168a->tdm_slots = slots;
-	pcm3168a->slot_width = slot_width;
-	pcm3168a->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
-	pcm3168a->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+	io_params->tdm_slots = slots;
+	io_params->slot_width = slot_width;
+	/* Ignore the not relevant mask for the DAI/direction */
+	if (dai->id == PCM3168A_DAI_DAC)
+		io_params->tdm_mask = tx_mask;
+	else
+		io_params->tdm_mask = rx_mask;
 
 	return 0;
 }
@@ -434,7 +431,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
-	bool tx, master_mode;
+	struct pcm3168a_io_params *io_params = &pcm3168a->io_params[dai->id];
+	bool master_mode;
 	u32 val, mask, shift, reg;
 	unsigned int rate, fmt, ratio, max_ratio;
 	unsigned int tdm_slots;
@@ -444,23 +442,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 
 	ratio = pcm3168a->sysclk / rate;
 
-	tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	if (tx) {
+	if (dai->id == PCM3168A_DAI_DAC) {
 		max_ratio = PCM3168A_NUM_SCKI_RATIOS_DAC;
 		reg = PCM3168A_DAC_PWR_MST_FMT;
 		mask = PCM3168A_DAC_MSDA_MASK;
 		shift = PCM3168A_DAC_MSDA_SHIFT;
-		master_mode = pcm3168a->dac_master_mode;
-		fmt = pcm3168a->dac_fmt;
 	} else {
 		max_ratio = PCM3168A_NUM_SCKI_RATIOS_ADC;
 		reg = PCM3168A_ADC_MST_FMT;
 		mask = PCM3168A_ADC_MSAD_MASK;
 		shift = PCM3168A_ADC_MSAD_SHIFT;
-		master_mode = pcm3168a->adc_master_mode;
-		fmt = pcm3168a->adc_fmt;
 	}
 
+	master_mode = io_params->master_mode;
+	fmt = io_params->fmt;
+
 	for (i = 0; i < max_ratio; i++) {
 		if (pcm3168a_scki_ratios[i] == ratio)
 			break;
@@ -471,8 +467,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	if (pcm3168a->slot_width)
-		slot_width = pcm3168a->slot_width;
+	if (io_params->slot_width)
+		slot_width = io_params->slot_width;
 	else
 		slot_width = params_width(params);
 
@@ -497,8 +493,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	if (pcm3168a->tdm_slots)
-		tdm_slots = pcm3168a->tdm_slots;
+	if (io_params->tdm_slots)
+		tdm_slots = io_params->tdm_slots;
 	else
 		tdm_slots = params_channels(params);
 
@@ -534,7 +530,7 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 
 	regmap_update_bits(pcm3168a->regmap, reg, mask, val);
 
-	if (tx) {
+	if (dai->id == PCM3168A_DAI_DAC) {
 		mask = PCM3168A_DAC_FMT_MASK;
 		shift = PCM3168A_DAC_FMT_SHIFT;
 	} else {
@@ -552,20 +548,13 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
-	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	unsigned int fmt;
 	unsigned int sample_min;
 	unsigned int channel_max;
 	unsigned int channel_maxs[] = {
-		6, /* rx */
-		8  /* tx */
+		8, /* DAC */
+		6  /* ADC */
 	};
 
-	if (tx)
-		fmt = pcm3168a->dac_fmt;
-	else
-		fmt = pcm3168a->adc_fmt;
-
 	/*
 	 * Available Data Bits
 	 *
@@ -578,7 +567,7 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream,
 	 * I2S
 	 * LEFT_J
 	 */
-	switch (fmt) {
+	switch (pcm3168a->io_params[dai->id].fmt) {
 	case PCM3168A_FMT_RIGHT_J:
 		sample_min  = 16;
 		channel_max =  2;
@@ -588,7 +577,7 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream,
 	case PCM3168A_FMT_DSP_A:
 	case PCM3168A_FMT_DSP_B:
 		sample_min  = 24;
-		channel_max = channel_maxs[tx];
+		channel_max = channel_maxs[dai->id];
 		break;
 	default:
 		sample_min  = 24;
@@ -600,8 +589,8 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream,
 				     sample_min, 32);
 
 	/* Allow all channels in multi DIN/DOUT mode */
-	if (pcm3168a->tdm_slots == 2)
-		channel_max = channel_maxs[tx];
+	if (pcm3168a->io_params[dai->id].tdm_slots == 2)
+		channel_max = channel_maxs[dai->id];
 
 	snd_pcm_hw_constraint_minmax(substream->runtime,
 				     SNDRV_PCM_HW_PARAM_CHANNELS,
@@ -609,26 +598,19 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream,
 
 	return 0;
 }
-static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
+static const struct snd_soc_dai_ops pcm3168a_dai_ops = {
 	.startup	= pcm3168a_startup,
-	.set_fmt	= pcm3168a_set_dai_fmt_dac,
+	.set_fmt	= pcm3168a_set_dai_fmt,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params,
 	.digital_mute	= pcm3168a_digital_mute,
 	.set_tdm_slot	= pcm3168a_set_tdm_slot,
 };
 
-static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = {
-	.startup	= pcm3168a_startup,
-	.set_fmt	= pcm3168a_set_dai_fmt_adc,
-	.set_sysclk	= pcm3168a_set_dai_sysclk,
-	.hw_params	= pcm3168a_hw_params,
-	.set_tdm_slot	= pcm3168a_set_tdm_slot,
-};
-
 static struct snd_soc_dai_driver pcm3168a_dais[] = {
 	{
 		.name = "pcm3168a-dac",
+		.id = PCM3168A_DAI_DAC,
 		.playback = {
 			.stream_name = "Playback",
 			.channels_min = 1,
@@ -636,10 +618,11 @@ static struct snd_soc_dai_driver pcm3168a_dais[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = PCM3168A_FORMATS
 		},
-		.ops = &pcm3168a_dac_dai_ops
+		.ops = &pcm3168a_dai_ops
 	},
 	{
 		.name = "pcm3168a-adc",
+		.id = PCM3168A_DAI_ADC,
 		.capture = {
 			.stream_name = "Capture",
 			.channels_min = 1,
@@ -647,7 +630,7 @@ static struct snd_soc_dai_driver pcm3168a_dais[] = {
 			.rates = SNDRV_PCM_RATE_8000_96000,
 			.formats = PCM3168A_FORMATS
 		},
-		.ops = &pcm3168a_adc_dai_ops
+		.ops = &pcm3168a_dai_ops
 	},
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 6fa4e0cae684d268d309d1b1f929d52e3df5649c Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Mon, 12 Aug 2019 12:52:26 +0300
Subject: ASoC: pcm3168a: Allow reconfiguration of tdm_slots and slot_width

When using right_j format and the codec is slave it can support 16bit
format, but only if slot_width == 16, in the same DAI mode the 24 bit
audio can work with 24 or 32 slot_width.

Because of this, the codec and CPU needs to be reconfigured when the sample
format changes.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190812095226.18870-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/pcm3168a.c | 12 ------------
 1 file changed, 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 75fa8e9ee894..50ed86d45c26 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -402,18 +402,6 @@ static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		return -EINVAL;
 	}
 
-	if (io_params->tdm_slots && io_params->tdm_slots != slots) {
-		dev_err(component->dev, "Not matching slots %d vs %d\n",
-			io_params->tdm_slots, slots);
-		return -EINVAL;
-	}
-
-	if (io_params->slot_width && io_params->slot_width != slot_width) {
-		dev_err(component->dev, "Not matching slot_width %d vs %d\n",
-			io_params->slot_width, slot_width);
-		return -EINVAL;
-	}
-
 	io_params->tdm_slots = slots;
 	io_params->slot_width = slot_width;
 	/* Ignore the not relevant mask for the DAI/direction */
-- 
cgit v1.2.3-59-g8ed1b


From 7da20788d3ff59506aebd946d8a424f01f2c1c66 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 12 Aug 2019 18:01:47 +0200
Subject: ALSA: hda: Set fifo_size for both playback and capture streams

Currently we set hdac_stream.fifo_size field only for the playback
stream by some odd reason I forgot, while this field isn't referred in
any places.  Actually this fifo_size field would have been required in
the position report correction for VIA chipset, but due to the lack of
the fifo_size set for capture streams, snd-hda-intel driver fetches
the register by itself.

This patch straightens and simplifies the code by setting the
fifo_size field for both playback and capture streams, and use it in
the HD-audio controller driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_stream.c   | 6 +-----
 sound/pci/hda/hda_intel.c | 8 +-------
 2 files changed, 2 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index fc68d4ce0a37..d8fe7ff0cd58 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -229,11 +229,7 @@ int snd_hdac_stream_setup(struct hdac_stream *azx_dev)
 	/* set the interrupt enable bits in the descriptor control register */
 	snd_hdac_stream_updatel(azx_dev, SD_CTL, 0, SD_INT_MASK);
 
-	if (azx_dev->direction == SNDRV_PCM_STREAM_PLAYBACK)
-		azx_dev->fifo_size =
-			snd_hdac_stream_readw(azx_dev, SD_FIFOSIZE) + 1;
-	else
-		azx_dev->fifo_size = 0;
+	azx_dev->fifo_size = snd_hdac_stream_readw(azx_dev, SD_FIFOSIZE) + 1;
 
 	/* when LPIB delay correction gives a small negative value,
 	 * we ignore it; currently set the threshold statically to
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3a209e07d5d8..dc8a83cb7393 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -84,8 +84,6 @@ enum {
 #define INTEL_SCH_HDA_DEVC      0x78
 #define INTEL_SCH_HDA_DEVC_NOSNOOP       (0x1<<11)
 
-/* Define IN stream 0 FIFO size offset in VIA controller */
-#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET	0x90
 /* Define VIA HD Audio Device ID*/
 #define VIA_HDAC_DEVICE_ID		0x3288
 
@@ -811,11 +809,7 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	mod_dma_pos = le32_to_cpu(*azx_dev->core.posbuf);
 	mod_dma_pos %= azx_dev->core.period_bytes;
 
-	/* azx_dev->fifo_size can't get FIFO size of in stream.
-	 * Get from base address + offset.
-	 */
-	fifo_size = readw(azx_bus(chip)->remap_addr +
-			  VIA_IN_STREAM0_FIFO_SIZE_OFFSET);
+	fifo_size = azx_stream(azx_dev)->fifo_size - 1;
 
 	if (azx_dev->insufficient) {
 		/* Link position never gather than FIFO size */
-- 
cgit v1.2.3-59-g8ed1b


From 68b953aeb50d9206de27d6c216e301f01dfac34d Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 12 Aug 2019 11:06:23 -0500
Subject: ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number

The SOF project maintains 6 topologies for HDaudio (iDisp or
HDaudio+iDisp, no DMIC, 2 DMICs, 4 DMICs). The user is currently
required to manually rename the topology file used in
/lib/firmware/intel/sof-tplg. We can do better to avoid such
renames and use logic to select the relevant file.

The NHLT information can be used to figure out which topology file
should be used.

Alternatively, when NHLT is not present in ACPI tables or is possibly
incorrect, a module parameter can provide that information, e.g. on
Up^2 board with the test DMIC kit.

Tested on Up^2 board and Acer Swift-SF314-55

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190812160623.20821-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/Kconfig |  1 +
 sound/soc/sof/intel/hda.c   | 75 +++++++++++++++++++++++++++++++++++++--------
 2 files changed, 64 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index dd14ce92fe10..545071afbe18 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -254,6 +254,7 @@ config SND_SOC_SOF_HDA
 	tristate
 	select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK
 	select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC
+	select SND_INTEL_NHLT
 	help
 	  This option is not user-selectable but automagically handled by
 	  'select' statements at a higher level
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 28eb780494aa..c97c004afa43 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -19,6 +19,7 @@
 #include <sound/hda_register.h>
 
 #include <linux/module.h>
+#include <sound/intel-nhlt.h>
 #include <sound/sof.h>
 #include <sound/sof/xtensa.h>
 #include "../ops.h"
@@ -49,6 +50,12 @@ module_param_named(use_msi, hda_use_msi, bool, 0444);
 MODULE_PARM_DESC(use_msi, "SOF HDA use PCI MSI mode");
 #endif
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+static int hda_dmic_num = -1;
+module_param_named(dmic_num, hda_dmic_num, int, 0444);
+MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number");
+#endif
+
 static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = {
 	{HDA_DSP_ROM_FW_MANIFEST_LOADED, "status: manifest loaded"},
 	{HDA_DSP_ROM_FW_FW_LOADED, "status: fw loaded"},
@@ -283,8 +290,26 @@ static int hda_init(struct snd_sof_dev *sdev)
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 
+static int check_nhlt_dmic(struct snd_sof_dev *sdev)
+{
+	struct nhlt_acpi_table *nhlt;
+	int dmic_num;
+
+	nhlt = intel_nhlt_init(sdev->dev);
+	if (nhlt) {
+		dmic_num = intel_nhlt_get_dmic_geo(sdev->dev, nhlt);
+		intel_nhlt_free(nhlt);
+		if (dmic_num == 2 || dmic_num == 4)
+			return dmic_num;
+	}
+
+	return 0;
+}
+
 static const char *fixup_tplg_name(struct snd_sof_dev *sdev,
-				   const char *sof_tplg_filename)
+				   const char *sof_tplg_filename,
+				   const char *idisp_str,
+				   const char *dmic_str)
 {
 	const char *tplg_filename = NULL;
 	char *filename;
@@ -298,7 +323,8 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev,
 	split_ext = strsep(&filename, ".");
 	if (split_ext) {
 		tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL,
-					       "%s-idisp.tplg", split_ext);
+					       "%s%s%s.tplg",
+					       split_ext, idisp_str, dmic_str);
 		if (!tplg_filename)
 			return NULL;
 	}
@@ -317,6 +343,9 @@ static int hda_init_caps(struct snd_sof_dev *sdev)
 	struct snd_sof_pdata *pdata = sdev->pdata;
 	struct snd_soc_acpi_mach *mach;
 	const char *tplg_filename;
+	const char *idisp_str;
+	const char *dmic_str;
+	int dmic_num;
 	int codec_num = 0;
 	int i;
 #endif
@@ -381,17 +410,39 @@ static int hda_init_caps(struct snd_sof_dev *sdev)
 			dev_info(bus->dev, "using HDA machine driver %s now\n",
 				 hda_mach->drv_name);
 
-			/* fixup topology file for HDMI only platforms */
-			if (codec_num == 1) {
-				/* use local variable for readability */
-				tplg_filename = pdata->tplg_filename;
-				tplg_filename = fixup_tplg_name(sdev, tplg_filename);
-				if (!tplg_filename) {
-					hda_codec_i915_exit(sdev);
-					return ret;
-				}
-				pdata->tplg_filename = tplg_filename;
+			if (codec_num == 1)
+				idisp_str = "-idisp";
+			else
+				idisp_str = "";
+
+			/* first check NHLT for DMICs */
+			dmic_num = check_nhlt_dmic(sdev);
+
+			/* allow for module parameter override */
+			if (hda_dmic_num != -1)
+				dmic_num = hda_dmic_num;
+
+			switch (dmic_num) {
+			case 2:
+				dmic_str = "-2ch";
+				break;
+			case 4:
+				dmic_str = "-4ch";
+				break;
+			default:
+				dmic_num = 0;
+				dmic_str = "";
+				break;
+			}
+
+			tplg_filename = pdata->tplg_filename;
+			tplg_filename = fixup_tplg_name(sdev, tplg_filename,
+							idisp_str, dmic_str);
+			if (!tplg_filename) {
+				hda_codec_i915_exit(sdev);
+				return ret;
 			}
+			pdata->tplg_filename = tplg_filename;
 		}
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From 7188f656cdf762d4ea8ce16b6aaf4c6b06e119ec Mon Sep 17 00:00:00 2001
From: Cheng-Yi Chiang <cychiang@chromium.org>
Date: Tue, 13 Aug 2019 15:44:30 +0800
Subject: ASoC: rockchip: rockchip_max98090: Set period size to 240

From stress testing of arecord, we found that period size
greater than ~900 will bring pl330 to DYING state and
can not recover within 100 iterations.
The result is that arecord will stuck and get I/O error,
and issue can not be recovered until reboot.

This issue does not happen when period size is small.
Set constraint of period size to 240 to prevent such issue.
With the constraint, there will be no issue after 2000 iterations.

We can revert this patch once the root cause is found
in rockchip's pl330 implementation.

Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190813074430.191791-1-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_max98090.c | 11 +++++++++++
 1 file changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c
index 782e534d4c0d..d54f672d38d8 100644
--- a/sound/soc/rockchip/rockchip_max98090.c
+++ b/sound/soc/rockchip/rockchip_max98090.c
@@ -138,8 +138,19 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int rk_aif1_startup(struct snd_pcm_substream *substream)
+{
+	/*
+	 * Set period size to 240 because pl330 has issue
+	 * dealing with larger period in stress testing.
+	 */
+	return snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 240, 240);
+}
+
 static const struct snd_soc_ops rk_aif1_ops = {
 	.hw_params = rk_aif1_hw_params,
+	.startup = rk_aif1_startup,
 };
 
 SND_SOC_DAILINK_DEFS(hifi,
-- 
cgit v1.2.3-59-g8ed1b


From ae3a5901dde2ab136ec0cebda2fccc48e810d2ec Mon Sep 17 00:00:00 2001
From: Masanari Iida <standby24x7@gmail.com>
Date: Tue, 13 Aug 2019 12:42:35 +0900
Subject: ASoC: ti: Fix typos in ti/Kconfig

This patch fixes some spelling typo in Kconfig.

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190813034235.30673-1-standby24x7@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/Kconfig | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index 2197f3e1eaed..87a9b9dd4e98 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -12,7 +12,7 @@ config SND_SOC_TI_SDMA_PCM
 
 comment "Texas Instruments DAI support for:"
 config SND_SOC_DAVINCI_ASP
-	tristate "daVinci Audio Serial Port (ASP) or McBSP suport"
+	tristate "daVinci Audio Serial Port (ASP) or McBSP support"
 	depends on ARCH_DAVINCI || COMPILE_TEST
 	select SND_SOC_TI_EDMA_PCM
 	help
@@ -33,7 +33,7 @@ config SND_SOC_DAVINCI_MCASP
 	  - Keystone devices
 
 config SND_SOC_DAVINCI_VCIF
-	tristate "daVinci Voice Interface (VCIF) suport"
+	tristate "daVinci Voice Interface (VCIF) support"
 	depends on ARCH_DAVINCI || COMPILE_TEST
 	select SND_SOC_TI_EDMA_PCM
 	help
-- 
cgit v1.2.3-59-g8ed1b


From 8da2d74c10fd71cc04e2585fe8bb3ced325f8e25 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 12 Aug 2019 09:03:05 -0500
Subject: ASoC: Intel: hsw: remove i386 build warning w/ size_t argument

Recent changes in the common IPC code introduced a build warning with
size_t fields, use the correct %zu format.

include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects
argument of type 'long unsigned int', but argument 4 has type 'size_t'
[-Wformat=]

Fixes: abf31feea26c0 ('ASoC: Intel: Update request-reply IPC model')
Reported-by: kbuild test robot <lkp@intel.com>
Cc: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-By: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190812140305.17570-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/haswell/sst-haswell-ipc.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index 5c73b11375e3..0ff89ea96ccf 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -2015,11 +2015,11 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw,
 
 	if (payload_size <= SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE) {
 		/* short parameter, mailbox can contain data */
-		dev_dbg(dev, "transfer parameter size : %lu\n",
+		dev_dbg(dev, "transfer parameter size : %zu\n",
 			request.size);
 
 		request.size = ALIGN(payload_size, 4);
-		dev_dbg(dev, "transfer parameter aligned size : %lu\n",
+		dev_dbg(dev, "transfer parameter aligned size : %zu\n",
 			request.size);
 
 		parameter = kzalloc(request.size, GFP_KERNEL);
-- 
cgit v1.2.3-59-g8ed1b


From 9c6c417d95d83999c16965186f6e755ad8a8b658 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 12 Aug 2019 14:05:02 -0500
Subject: ASoC: SOF: fix HDA direct MMIO access

The recent change to remove the bus->io_ops callbacks used an older
version of the SOF code base, and when merged into Mark's for-next it
invalidated changes, resulting in broken compilation identified by
kbuild and reproduced during the weekly SOF rebase.

Restore SOF code overridden by git merge and apply Takashi's intended
change in the 'right' location.

Fixes: c2f16a94a8049 ("Merge branch 'topic/hda-bus-ops-cleanup'")
Reported-by: kbuild test robot <lkp@intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190812190502.30729-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-ctrl.c |  2 +-
 sound/soc/sof/intel/hda-dsp.c  | 39 ---------------------------------------
 2 files changed, 1 insertion(+), 40 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index a7fee403cb90..bc41028a7a01 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -254,7 +254,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	/* Reset stream-to-link mapping */
 	list_for_each_entry(hlink, &bus->hlink_list, list)
-		bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
+		writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
 #endif
 
 	bus->chip_init = true;
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 097727cda5cb..fb55a3c5afd0 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -354,45 +354,6 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume)
 		return ret;
 	}
 
-	hda_dsp_ctrl_misc_clock_gating(sdev, false);
-
-	/* Reset stream-to-link mapping */
-	list_for_each_entry(hlink, &bus->hlink_list, list)
-		writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
-
-	hda_dsp_ctrl_misc_clock_gating(sdev, true);
-#else
-
-	hda_dsp_ctrl_misc_clock_gating(sdev, false);
-
-	/* reset controller */
-	ret = hda_dsp_ctrl_link_reset(sdev, true);
-	if (ret < 0) {
-		dev_err(sdev->dev,
-			"error: failed to reset controller during resume\n");
-		return ret;
-	}
-
-	/* take controller out of reset */
-	ret = hda_dsp_ctrl_link_reset(sdev, false);
-	if (ret < 0) {
-		dev_err(sdev->dev,
-			"error: failed to ready controller during resume\n");
-		return ret;
-	}
-
-	/* enable hda bus irq */
-	snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL,
-				SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN,
-				SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN);
-
-	hda_dsp_ctrl_misc_clock_gating(sdev, true);
-#endif
-
-	/* enable ppcap interrupt */
-	hda_dsp_ctrl_ppcap_enable(sdev, true);
-	hda_dsp_ctrl_ppcap_int_enable(sdev, true);
-
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	/* check jack status */
 	if (runtime_resume)
-- 
cgit v1.2.3-59-g8ed1b


From 31fa571c01d869c659d6bc0a7960a55ce314923c Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Tue, 13 Aug 2019 22:43:23 +0800
Subject: ALSA: hda/sigmatel - remove unused variable 'stac9200_core_init'

sound/pci/hda/patch_sigmatel.c:981:30: warning:
 stac9200_core_init defined but not used [-Wunused-const-variable=]

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 9 ---------
 1 file changed, 9 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 0d9b62768241..894f3f509e76 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -975,15 +975,6 @@ static int stac_create_spdif_mux_ctls(struct hda_codec *codec)
 	return 0;
 }
 
-/*
- */
-
-static const struct hda_verb stac9200_core_init[] = {
-	/* set dac0mux for dac converter */
-	{ 0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{}
-};
-
 static const struct hda_verb stac9200_eapd_init[] = {
 	/* set dac0mux for dac converter */
 	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-- 
cgit v1.2.3-59-g8ed1b


From d7da429339f503854b777ed941456841a72b48b6 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Tue, 13 Aug 2019 15:01:51 +0100
Subject: ALSA: sb: remove redundant assignment to variable result

Variable result is initialized to a value that is never read and it is
re-assigned later. The initialization is redundant and can be removed.

Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/isa/sb/sb_common.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
index 162338f1b68a..ff031d670400 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -80,7 +80,7 @@ int snd_sbdsp_reset(struct snd_sb *chip)
 
 static int snd_sbdsp_version(struct snd_sb * chip)
 {
-	unsigned int result = -ENODEV;
+	unsigned int result;
 
 	snd_sbdsp_command(chip, SB_DSP_GET_VERSION);
 	result = (short) snd_sbdsp_get_byte(chip) << 8;
-- 
cgit v1.2.3-59-g8ed1b


From f2dbe87c5ac1f88e6007ba1f1374f4bd8a197fb6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Aug 2019 17:11:28 +0200
Subject: ALSA: hda - Drop unsol event handler for Intel HDMI codecs

We don't need to deal with the unsol events for Intel chips that are
tied with the graphics via audio component notifier.  Although the
presence of the audio component is checked at the beginning of
hdmi_unsol_event(), better to short cut by dropping unsol_event ops.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204565
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2096993eaf28..933c7bf47ef6 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2760,6 +2760,8 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
 /* precondition and allocation for Intel codecs */
 static int alloc_intel_hdmi(struct hda_codec *codec)
 {
+	int err;
+
 	/* requires i915 binding */
 	if (!codec->bus->core.audio_component) {
 		codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
@@ -2768,7 +2770,12 @@ static int alloc_intel_hdmi(struct hda_codec *codec)
 		return -ENODEV;
 	}
 
-	return alloc_generic_hdmi(codec);
+	err = alloc_generic_hdmi(codec);
+	if (err < 0)
+		return err;
+	/* no need to handle unsol events */
+	codec->patch_ops.unsol_event = NULL;
+	return 0;
 }
 
 /* parse and post-process for Intel codecs */
-- 
cgit v1.2.3-59-g8ed1b


From ee5f85d9290fe25d460bd320b7fe073075d72d33 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Aug 2019 17:25:39 +0200
Subject: ALSA: hda: Add codec on bus address table lately

The call of snd_hdac_bus_add_device() is needed only for registering
the codec onto the bus caddr_tbl[] that is referred essentially only
in the unsol event handler.  That is, the reason of this call and the
release by the counter-part function snd_hdac_bus_remove_device() is
just to assure that the unsol event gets notified to the codec.

But the current implementation of the unsol notification wouldn't work
properly when the codec is still in a premature init state.  So this
patch tries to work around it by delaying the caddr_tbl[] registration
at the point of snd_hdac_device_register().

Also, the order of snd_hdac_bus_remove_device() and device_del() calls
are shuffled to make sure that the unsol event is masked before
deleting the device.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204565
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_device.c | 21 ++++++++++++---------
 1 file changed, 12 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index b26cc93e7e10..033bcef8751a 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -61,10 +61,6 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus,
 	pm_runtime_get_noresume(&codec->dev);
 	atomic_set(&codec->in_pm, 0);
 
-	err = snd_hdac_bus_add_device(bus, codec);
-	if (err < 0)
-		goto error;
-
 	/* fill parameters */
 	codec->vendor_id = snd_hdac_read_parm(codec, AC_NODE_ROOT,
 					      AC_PAR_VENDOR_ID);
@@ -143,15 +139,22 @@ int snd_hdac_device_register(struct hdac_device *codec)
 	err = device_add(&codec->dev);
 	if (err < 0)
 		return err;
+	err = snd_hdac_bus_add_device(codec->bus, codec);
+	if (err < 0)
+		goto error;
 	mutex_lock(&codec->widget_lock);
 	err = hda_widget_sysfs_init(codec);
 	mutex_unlock(&codec->widget_lock);
-	if (err < 0) {
-		device_del(&codec->dev);
-		return err;
-	}
+	if (err < 0)
+		goto error_remove;
 
 	return 0;
+
+ error_remove:
+	snd_hdac_bus_remove_device(codec->bus, codec);
+ error:
+	device_del(&codec->dev);
+	return err;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_device_register);
 
@@ -165,8 +168,8 @@ void snd_hdac_device_unregister(struct hdac_device *codec)
 		mutex_lock(&codec->widget_lock);
 		hda_widget_sysfs_exit(codec);
 		mutex_unlock(&codec->widget_lock);
-		device_del(&codec->dev);
 		snd_hdac_bus_remove_device(codec->bus, codec);
+		device_del(&codec->dev);
 	}
 }
 EXPORT_SYMBOL_GPL(snd_hdac_device_unregister);
-- 
cgit v1.2.3-59-g8ed1b


From 53eff75e5f4dd4b9bc489955fdc60fde48d85e93 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 14 Aug 2019 18:27:08 +0200
Subject: ALSA: hda: Drop export of snd_hdac_bus_add/remove_device()

snd_hdac_bus_add_device() and snd_hdac_remove_device() are called only
internally in hda-core.  Let's drop the exports of them and move the
declarations into local.h.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hdaudio.h | 3 ---
 sound/hda/hdac_bus.c    | 3 +--
 sound/hda/local.h       | 4 ++++
 3 files changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 4af4af55e854..edb176a265c7 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -374,9 +374,6 @@ int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr,
 				    unsigned int cmd, unsigned int *res);
 void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex);
 
-int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec);
-void snd_hdac_bus_remove_device(struct hdac_bus *bus,
-				struct hdac_device *codec);
 void snd_hdac_bus_process_unsol_events(struct work_struct *work);
 
 static inline void snd_hdac_codec_link_up(struct hdac_device *codec)
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index cd25e2b3f7f2..18ed3185df82 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -9,6 +9,7 @@
 #include <linux/module.h>
 #include <linux/export.h>
 #include <sound/hdaudio.h>
+#include "local.h"
 #include "trace.h"
 
 static const struct hdac_bus_ops default_ops = {
@@ -196,7 +197,6 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec)
 	bus->num_codecs++;
 	return 0;
 }
-EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device);
 
 /**
  * snd_hdac_bus_remove_device - Remove a codec from bus
@@ -215,7 +215,6 @@ void snd_hdac_bus_remove_device(struct hdac_bus *bus,
 	bus->num_codecs--;
 	flush_work(&bus->unsol_work);
 }
-EXPORT_SYMBOL_GPL(snd_hdac_bus_remove_device);
 
 #ifdef CONFIG_SND_HDA_ALIGNED_MMIO
 /* Helpers for aligned read/write of mmio space, for Tegra */
diff --git a/sound/hda/local.h b/sound/hda/local.h
index 877631e39373..3a4e303169a6 100644
--- a/sound/hda/local.h
+++ b/sound/hda/local.h
@@ -33,4 +33,8 @@ int hda_widget_sysfs_reinit(struct hdac_device *codec, hda_nid_t start_nid,
 			    int num_nodes);
 void hda_widget_sysfs_exit(struct hdac_device *codec);
 
+int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec);
+void snd_hdac_bus_remove_device(struct hdac_bus *bus,
+				struct hdac_device *codec);
+
 #endif /* __HDAC_LOCAL_H */
-- 
cgit v1.2.3-59-g8ed1b


From ddf7cb83b0f45feb94ad89a987f600c766c463ca Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 14 Aug 2019 19:59:44 +0200
Subject: ALSA: hda: Unexport a few more stuff

Drop EXPORT_SYMBOL*() from a few more stuff in HD-audio core that
aren't used outside.  Particular the unsol event handler can be
staticized now because the recent change removed all external
callers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hdaudio.h | 6 ------
 sound/hda/hdac_bus.c    | 5 +++--
 sound/hda/hdac_device.c | 6 ++----
 sound/hda/hdac_regmap.c | 1 +
 sound/hda/local.h       | 3 +++
 5 files changed, 9 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index edb176a265c7..b260c5fd2337 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -122,10 +122,6 @@ int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size);
 
 int snd_hdac_refresh_widgets(struct hdac_device *codec);
 
-unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
-			       unsigned int verb, unsigned int parm);
-int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd,
-		       unsigned int flags, unsigned int *res);
 int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid,
 		  unsigned int verb, unsigned int parm, unsigned int *res);
 int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm,
@@ -374,8 +370,6 @@ int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr,
 				    unsigned int cmd, unsigned int *res);
 void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex);
 
-void snd_hdac_bus_process_unsol_events(struct work_struct *work);
-
 static inline void snd_hdac_codec_link_up(struct hdac_device *codec)
 {
 	set_bit(codec->addr, &codec->bus->codec_powered);
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 18ed3185df82..8f19876244eb 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -12,6 +12,8 @@
 #include "local.h"
 #include "trace.h"
 
+static void snd_hdac_bus_process_unsol_events(struct work_struct *work);
+
 static const struct hdac_bus_ops default_ops = {
 	.command = snd_hdac_bus_send_cmd,
 	.get_response = snd_hdac_bus_get_response,
@@ -149,7 +151,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_queue_event);
 /*
  * process queued unsolicited events
  */
-void snd_hdac_bus_process_unsol_events(struct work_struct *work)
+static void snd_hdac_bus_process_unsol_events(struct work_struct *work)
 {
 	struct hdac_bus *bus = container_of(work, struct hdac_bus, unsol_work);
 	struct hdac_device *codec;
@@ -172,7 +174,6 @@ void snd_hdac_bus_process_unsol_events(struct work_struct *work)
 			drv->unsol_event(codec, res);
 	}
 }
-EXPORT_SYMBOL_GPL(snd_hdac_bus_process_unsol_events);
 
 /**
  * snd_hdac_bus_add_device - Add a codec to bus
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 033bcef8751a..bf83d7062ef6 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -221,8 +221,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);
  *
  * Return an encoded command verb or -1 for error.
  */
-unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
-			       unsigned int verb, unsigned int parm)
+static unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
+				      unsigned int verb, unsigned int parm)
 {
 	u32 val, addr;
 
@@ -240,7 +240,6 @@ unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
 	val |= parm;
 	return val;
 }
-EXPORT_SYMBOL_GPL(snd_hdac_make_cmd);
 
 /**
  * snd_hdac_exec_verb - execute an encoded verb
@@ -261,7 +260,6 @@ int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd,
 		return codec->exec_verb(codec, cmd, flags, res);
 	return snd_hdac_bus_exec_verb(codec->bus, codec->addr, cmd, res);
 }
-EXPORT_SYMBOL_GPL(snd_hdac_exec_verb);
 
 
 /**
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index f399a1552e73..286361ecd640 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -21,6 +21,7 @@
 #include <sound/core.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_regmap.h>
+#include "local.h"
 
 static int codec_pm_lock(struct hdac_device *codec)
 {
diff --git a/sound/hda/local.h b/sound/hda/local.h
index 3a4e303169a6..5b935219352f 100644
--- a/sound/hda/local.h
+++ b/sound/hda/local.h
@@ -37,4 +37,7 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec);
 void snd_hdac_bus_remove_device(struct hdac_bus *bus,
 				struct hdac_device *codec);
 
+int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd,
+		       unsigned int flags, unsigned int *res);
+
 #endif /* __HDAC_LOCAL_H */
-- 
cgit v1.2.3-59-g8ed1b


From 630742c296341a8cfe00dfd941392025ba8dd4e8 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:23:00 +0800
Subject: ASoC: es8328: Fix copy-paste error in es8328_right_line_controls

It seems 'es8328_rline_enum' should be used
in es8328_right_line_controls

Fixes: 567e4f98922c ("ASoC: add es8328 codec driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8328.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 822a25a8f53c..69b81e704127 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -228,7 +228,7 @@ static const struct soc_enum es8328_rline_enum =
 			      ARRAY_SIZE(es8328_line_texts),
 			      es8328_line_texts);
 static const struct snd_kcontrol_new es8328_right_line_controls =
-	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+	SOC_DAPM_ENUM("Route", es8328_rline_enum);
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
-- 
cgit v1.2.3-59-g8ed1b


From 9b4275c415acca6264a3d7f1182589959c93d530 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:01:57 +0800
Subject: ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'

sound/soc/codecs/cs4349.c:358:32: warning:
 cs4349_runtime_pm defined but not used [-Wunused-const-variable=]

cs4349_runtime_pm ops already defined, it seems
we should enable it.

Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs4349.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index 09716fab1e26..3381209a882d 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -378,6 +378,7 @@ static struct i2c_driver cs4349_i2c_driver = {
 	.driver = {
 		.name		= "cs4349",
 		.of_match_table	= cs4349_of_match,
+		.pm = &cs4349_runtime_pm,
 	},
 	.id_table	= cs4349_i2c_id,
 	.probe		= cs4349_i2c_probe,
-- 
cgit v1.2.3-59-g8ed1b


From 554b75bde64bcad9662530726d1483f7ef012069 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:19:20 +0800
Subject: ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls

sound/soc/codecs/wm8737.c:112:29: warning:
 high_3d defined but not used [-Wunused-const-variable=]

'high_3d' should be used for 3D High Cut-off.

Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190815091920.64480-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8737.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 0c246fb5e5ac..7a3f9fbe8d53 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -167,7 +167,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0),
 SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0),
 SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0),
 SOC_ENUM("3D Low Cut-off", low_3d),
-SOC_ENUM("3D High Cut-off", low_3d),
+SOC_ENUM("3D High Cut-off", high_3d),
 SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv),
 
 SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0),
-- 
cgit v1.2.3-59-g8ed1b


From 0faf1237c60a3791d7ff32035d3097d3e022e68f Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Tue, 13 Aug 2019 22:25:01 +0800
Subject: ASoC: soc-core: Fix -Wunused-const-variable warning

If CONFIG_DMI is not set, gcc warns:

sound/soc/soc-core.c:81:27: warning:
 dmi_blacklist defined but not used [-Wunused-const-variable=]

Add #ifdef guard around it.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190813142501.13080-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e9f44505cc3e..abe2f47cee6e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -73,6 +73,7 @@ static int pmdown_time = 5000;
 module_param(pmdown_time, int, 0);
 MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
 
+#ifdef CONFIG_DMI
 /*
  * If a DMI filed contain strings in this blacklist (e.g.
  * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken
@@ -87,6 +88,7 @@ static const char * const dmi_blacklist[] = {
 	"Board Product Name",
 	NULL,	/* terminator */
 };
+#endif
 
 static ssize_t pmdown_time_show(struct device *dev,
 				struct device_attribute *attr, char *buf)
-- 
cgit v1.2.3-59-g8ed1b


From a9e792d006edbd33724f2eb858887d3b591d82c5 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Tue, 13 Aug 2019 22:38:11 +0800
Subject: ASoC: mt2701: remove unused variables

sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:799:38: warning:
 mt2701_afe_o23_mix defined but not used [-Wunused-const-variable=]
sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:803:38: warning:
 mt2701_afe_o24_mix defined but not used [-Wunused-const-variable=]
sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:835:38: warning:
 mt2701_afe_multi_ch_out_i2s4 defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190813143811.31456-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 13 -------------
 1 file changed, 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
index 9af76ae315a5..d7f5defa50c2 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
@@ -796,14 +796,6 @@ static const struct snd_kcontrol_new mt2701_afe_o22_mix[] = {
 	SOC_DAPM_SINGLE_AUTODISABLE("I19 Switch", AFE_CONN22, 19, 1, 0),
 };
 
-static const struct snd_kcontrol_new mt2701_afe_o23_mix[] = {
-	SOC_DAPM_SINGLE_AUTODISABLE("I20 Switch", AFE_CONN23, 20, 1, 0),
-};
-
-static const struct snd_kcontrol_new mt2701_afe_o24_mix[] = {
-	SOC_DAPM_SINGLE_AUTODISABLE("I21 Switch", AFE_CONN24, 21, 1, 0),
-};
-
 static const struct snd_kcontrol_new mt2701_afe_o31_mix[] = {
 	SOC_DAPM_SINGLE_AUTODISABLE("I35 Switch", AFE_CONN41, 9, 1, 0),
 };
@@ -832,11 +824,6 @@ static const struct snd_kcontrol_new mt2701_afe_multi_ch_out_i2s3[] = {
 				    PWR2_TOP_CON, 18, 1, 0),
 };
 
-static const struct snd_kcontrol_new mt2701_afe_multi_ch_out_i2s4[] = {
-	SOC_DAPM_SINGLE_AUTODISABLE("Multich I2S4 Out Switch",
-				    PWR2_TOP_CON, 19, 1, 0),
-};
-
 static const struct snd_soc_dapm_widget mt2701_afe_pcm_widgets[] = {
 	/* inter-connections */
 	SND_SOC_DAPM_MIXER("I00", SND_SOC_NOPM, 0, 0, NULL, 0),
-- 
cgit v1.2.3-59-g8ed1b


From 57c3ed42f52cdc51f416c93b19708ef6ceb4a00b Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Tue, 13 Aug 2019 22:39:52 +0800
Subject: ASoC: mediatek: mt8183-da7219-max98357: remove unused variable

sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:120:1: warning:
 mt8183_da7219_max98357_dapm_widgets defined but not used [-Wunused-const-variable=]
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:124:40: warning:
 mt8183_da7219_max98357_dapm_routes defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190813143952.29232-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 9 ---------
 1 file changed, 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index 2a6097174614..43f99e59a078 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -116,15 +116,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static const struct snd_soc_dapm_widget
-mt8183_da7219_max98357_dapm_widgets[] = {
-	SND_SOC_DAPM_OUTPUT("IT6505_8CH"),
-};
-
-static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = {
-	{"IT6505_8CH", NULL, "TDM"},
-};
-
 /* FE */
 SND_SOC_DAILINK_DEFS(playback1,
 	DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
-- 
cgit v1.2.3-59-g8ed1b


From d59170b42610c7cbc6e96431ca8357a8bdbf592b Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Tue, 13 Aug 2019 22:41:22 +0800
Subject: ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: remove unused
 variables

sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:50:1: warning:
 mt8183_mt6358_ts3a227_max98357_dapm_widgets defined but not used [-Wunused-const-variable=]
sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:55:1: warning:
 mt8183_mt6358_ts3a227_max98357_dapm_routes defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190813144122.67676-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 10 ----------
 1 file changed, 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 53f54078f78c..272766c1b859 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -46,16 +46,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static const struct snd_soc_dapm_widget
-mt8183_mt6358_ts3a227_max98357_dapm_widgets[] = {
-	SND_SOC_DAPM_OUTPUT("IT6505_8CH"),
-};
-
-static const struct snd_soc_dapm_route
-mt8183_mt6358_ts3a227_max98357_dapm_routes[] = {
-	{"IT6505_8CH", NULL, "TDM"},
-};
-
 static int
 mt8183_mt6358_ts3a227_max98357_bt_sco_startup(
 	struct snd_pcm_substream *substream)
-- 
cgit v1.2.3-59-g8ed1b


From ccb1fa21ef58a2ac15519bb878470762e967e8b3 Mon Sep 17 00:00:00 2001
From: Jiaxin Yu <jiaxin.yu@mediatek.com>
Date: Mon, 5 Aug 2019 12:56:37 +0800
Subject: ASoC: mediatek: mt6358: add delay after dmic clock on

Most dmics produce a high level when they receive clock. The difference
between power-on and memory record time is about 10ms, but the dmic
needs 50ms to output normal data.

This commit add 100ms delay after SoC output clock so that we can cut
off the pop noise at the beginning.

Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1564980997-11359-1-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/mt6358.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c
index c17250aab2d0..bb737fd678cc 100644
--- a/sound/soc/codecs/mt6358.c
+++ b/sound/soc/codecs/mt6358.c
@@ -1730,6 +1730,10 @@ static int mt6358_dmic_enable(struct mt6358_priv *priv)
 
 	/* UL turn on */
 	regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_L, 0x0003);
+
+	/* Prevent pop noise form dmic hw */
+	msleep(100);
+
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From ff30779bd50eb823e2e75cfc7cc8a3373bfa3fd9 Mon Sep 17 00:00:00 2001
From: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Date: Fri, 21 Jun 2019 14:31:16 +0300
Subject: ASoC: Intel: Skylake: Print constant literals from format specifier

Instead of using two additional "%s" specifiers, put the constant string
literals directly to the format specifier.

Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/20190621113116.47525-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-sst.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 8af7546def1f..61a8e4756a2b 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -413,8 +413,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid)
 	int ret = 0;
 	char mod_name[64]; /* guid str = 32 chars + 4 hyphens */
 
-	snprintf(mod_name, sizeof(mod_name), "%s%pUL%s",
-					     "intel/dsp_fw_", guid, ".bin");
+	snprintf(mod_name, sizeof(mod_name), "intel/dsp_fw_%pUL.bin", guid);
 
 	module_entry = skl_module_get_from_id(ctx, mod_id);
 	if (module_entry == NULL) {
-- 
cgit v1.2.3-59-g8ed1b


From c25b456dc579298ac0ed7304f7d06a66288e96df Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 16:54:54 +0800
Subject: ASoC: cs42l73: remove unused variables 'vsp_output_mux' and
 'xsp_output_mux'

sound/soc/codecs/cs42l73.c:276:38: warning:
 vsp_output_mux defined but not used [-Wunused-const-variable=]
sound/soc/codecs/cs42l73.c:279:38: warning:
 xsp_output_mux defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815085454.30384-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l73.c | 6 ------
 1 file changed, 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index a81739367109..36089f8bcf0a 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -273,12 +273,6 @@ static SOC_ENUM_SINGLE_DECL(xsp_output_mux_enum,
 			    CS42L73_MIXERCTL, 4,
 			    cs42l73_spo_mixer_text);
 
-static const struct snd_kcontrol_new vsp_output_mux =
-	SOC_DAPM_ENUM("Route", vsp_output_mux_enum);
-
-static const struct snd_kcontrol_new xsp_output_mux =
-	SOC_DAPM_ENUM("Route", xsp_output_mux_enum);
-
 static const struct snd_kcontrol_new hp_amp_ctl =
 	SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1);
 
-- 
cgit v1.2.3-59-g8ed1b


From bed7f1469f08fd123cdec7a351ef0d875feadcf6 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:01:57 +0800
Subject: ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'

sound/soc/codecs/cs4349.c:358:32: warning:
 cs4349_runtime_pm defined but not used [-Wunused-const-variable=]

cs4349_runtime_pm ops already defined, it seems
we should enable it.

Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs4349.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index 09716fab1e26..3381209a882d 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -378,6 +378,7 @@ static struct i2c_driver cs4349_i2c_driver = {
 	.driver = {
 		.name		= "cs4349",
 		.of_match_table	= cs4349_of_match,
+		.pm = &cs4349_runtime_pm,
 	},
 	.id_table	= cs4349_i2c_id,
 	.probe		= cs4349_i2c_probe,
-- 
cgit v1.2.3-59-g8ed1b


From 9d22142c9b1ec8612b880121dd0bc27311cbb2b5 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:04:04 +0800
Subject: ASoC: max98371: remove unused variable 'max98371_noload_gain_tlv'

sound/soc/codecs/max98371.c:157:35: warning:
 max98371_noload_gain_tlv defined but not used [-Wunused-const-variable=]

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815090404.72752-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98371.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
index ce801489a86d..dfee05f985bd 100644
--- a/sound/soc/codecs/max98371.c
+++ b/sound/soc/codecs/max98371.c
@@ -154,10 +154,6 @@ static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv,
 	8, 10, TLV_DB_SCALE_ITEM(400, 100, 0)
 );
 
-static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv,
-	0, 11, TLV_DB_SCALE_ITEM(950, 100, 0),
-);
-
 static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1);
 
 static const struct snd_kcontrol_new max98371_snd_controls[] = {
-- 
cgit v1.2.3-59-g8ed1b


From 5b366753c1c12feead0ae53b45482f569ed5399c Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:06:02 +0800
Subject: ASoC: rt1011: remove unused variable 'dac_vol_tlv' and 'adc_vol_tlv'

sound/soc/codecs/rt1011.c:981:35: warning:
 dac_vol_tlv defined but not used [-Wunused-const-variable=]
sound/soc/codecs/rt1011.c:982:35: warning:
 adc_vol_tlv defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815090602.9000-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1011.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 638abcaf52b3..fa34565a3938 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -978,9 +978,6 @@ static bool rt1011_readable_register(struct device *dev, unsigned int reg)
 	}
 }
 
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9435, 37, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1739, 37, 0);
-
 static const char * const rt1011_din_source_select[] = {
 	"Left",
 	"Right",
-- 
cgit v1.2.3-59-g8ed1b


From ab0ac2707784a966927c229752849c343bd1dbbf Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:15:34 +0800
Subject: ASoC: tlv320aic23: remove unused variable 'tlv320aic23_rec_src'

sound/soc/codecs/tlv320aic23.c:70:29: warning:
 tlv320aic23_rec_src defined but not used [-Wunused-const-variable=]

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815091534.57780-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic23.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 080a840c987a..f8e2f4b74db3 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -67,8 +67,6 @@ static SOC_ENUM_SINGLE_DECL(rec_src_enum,
 static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
 SOC_DAPM_ENUM("Input Select", rec_src_enum);
 
-static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src,
-			    TLV320AIC23_ANLG, 2, rec_src_text);
 static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph,
 			    TLV320AIC23_DIGT, 1, deemph_text);
 
-- 
cgit v1.2.3-59-g8ed1b


From 6d6376b143d59ab1b8635807c78d224d03580418 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:17:38 +0800
Subject: ASoC: tlv320aic31xx: remove unused variable 'cm_m_enum'

sound/soc/codecs/tlv320aic31xx.c:261:29: warning:
 cm_m_enum defined but not used [-Wunused-const-variable=]

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815091738.21680-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic31xx.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 26a4f6cd3288..df627a08def9 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -258,7 +258,6 @@ static SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4,
 static SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2,
 	mic_select_text);
 
-static SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
 static SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4,
 	mic_select_text);
 
-- 
cgit v1.2.3-59-g8ed1b


From 97d39be9ce5befc2e36f15d0df33832e0f633565 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:20:56 +0800
Subject: ASoC: es8328: remove unused variable 'pga_tlv'

sound/soc/codecs/es8328.c:102:35: warning:
 pga_tlv defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815092056.28724-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8328.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 822a25a8f53c..4a3d303fedfb 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -99,7 +99,6 @@ static SOC_ENUM_SINGLE_DECL(adcpol,
 
 static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0);
 static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0);
-static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
 static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0);
 
-- 
cgit v1.2.3-59-g8ed1b


From d63887bc4f50fede7013bda7c733d58ecc43efc1 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:23:00 +0800
Subject: ASoC: es8328: Fix copy-paste error in es8328_right_line_controls

It seems 'es8328_rline_enum' should be used
in es8328_right_line_controls

Fixes: 567e4f98922c ("ASoC: add es8328 codec driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8328.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 4a3d303fedfb..fdf64c29f563 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -227,7 +227,7 @@ static const struct soc_enum es8328_rline_enum =
 			      ARRAY_SIZE(es8328_line_texts),
 			      es8328_line_texts);
 static const struct snd_kcontrol_new es8328_right_line_controls =
-	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+	SOC_DAPM_ENUM("Route", es8328_rline_enum);
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
-- 
cgit v1.2.3-59-g8ed1b


From e33d565795930d0341bb946aec457a814ccd53e6 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:24:36 +0800
Subject: ASoC: cs42l56: remove unused variable 'adc_swap_enum'

sound/soc/codecs/cs42l56.c:206:30: warning:
 adc_swap_enum defined but not used [-Wunused-const-variable=]

It is never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815092436.34632-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l56.c | 8 --------
 1 file changed, 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index b4d7627525f9..ac569ab3d30f 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -199,14 +199,6 @@ static const struct soc_enum beep_bass_enum =
 	SOC_ENUM_SINGLE(CS42L56_BEEP_TONE_CFG, 1,
 			ARRAY_SIZE(beep_bass_text), beep_bass_text);
 
-static const char * const adc_swap_text[] = {
-	"None", "A+B/2", "A-B/2", "Swap"
-};
-
-static const struct soc_enum adc_swap_enum =
-	SOC_ENUM_SINGLE(CS42L56_MISC_ADC_CTL, 3,
-			ARRAY_SIZE(adc_swap_text), adc_swap_text);
-
 static const char * const pgaa_mux_text[] = {
 	"AIN1A", "AIN2A", "AIN3A"};
 
-- 
cgit v1.2.3-59-g8ed1b


From 12f0bfadf69bb154052722e7e4e5cd1639044c76 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 15 Aug 2019 17:25:47 +0800
Subject: ASoC: 88pm860x: remove unused variables 'pcm_switch_controls' and
 'aif1_mux'

sound/soc/codecs/88pm860x-codec.c:533:38: warning:
 pcm_switch_controls defined but not used [-Wunused-const-variable=]
sound/soc/codecs/88pm860x-codec.c:560:38: warning:
 aif1_mux defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815092547.29564-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/88pm860x-codec.c | 15 ---------------
 1 file changed, 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index e982722b448e..00b2c43d28a1 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -529,10 +529,6 @@ static const struct snd_kcontrol_new pm860x_snd_controls[] = {
  * DAPM Controls
  */
 
-/* PCM Switch / PCM Interface */
-static const struct snd_kcontrol_new pcm_switch_controls =
-	SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0);
-
 /* AUX1 Switch */
 static const struct snd_kcontrol_new aux1_switch_controls =
 	SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0);
@@ -549,17 +545,6 @@ static const struct snd_kcontrol_new lepa_switch_controls =
 static const struct snd_kcontrol_new repa_switch_controls =
 	SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0);
 
-/* PCM Mux / Mux7 */
-static const char *aif1_text[] = {
-	"PCM L", "PCM R",
-};
-
-static SOC_ENUM_SINGLE_DECL(aif1_enum,
-			    PM860X_PCM_IFACE_3, 6, aif1_text);
-
-static const struct snd_kcontrol_new aif1_mux =
-	SOC_DAPM_ENUM("PCM Mux", aif1_enum);
-
 /* I2S Mux / Mux9 */
 static const char *i2s_din_text[] = {
 	"DIN", "DIN1",
-- 
cgit v1.2.3-59-g8ed1b


From 72b745e3ad65deac94ea4eb83262c52ba3ffdb5b Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Tue, 13 Aug 2019 13:45:32 +0300
Subject: ASoC: core: Move pcm_mutex up to card level from snd_soc_pcm_runtime

The pcm_mutex is used to prevent concurrent execution of snd_pcm_ops
callbacks. This works fine most of the cases but it can not handle setups
when the same DAI is used by different rtd, for example:
pcm3168a have two DAIs: one for Playback and one for Capture.
If the codec is connected to a single CPU DAI we need to have two dai_link
to support both playback and capture.

In this case the snd_pcm_ops callbacks can be executed in parallel causing
unexpected races in DAI drivers.

By moving the pcm_mutex up to card level this can be solved
while - hopefully - not breaking other setups.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190813104532.16669-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h      |  6 ++++--
 sound/soc/soc-compress.c | 48 ++++++++++++++++++++++++------------------------
 sound/soc/soc-core.c     |  2 +-
 sound/soc/soc-pcm.c      | 36 ++++++++++++++++++------------------
 4 files changed, 47 insertions(+), 45 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index b1fe5ebea257..5c841c2ee814 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -988,6 +988,10 @@ struct snd_soc_card {
 	struct mutex mutex;
 	struct mutex dapm_mutex;
 
+	/* Mutex for PCM operations */
+	struct mutex pcm_mutex;
+	enum snd_soc_pcm_subclass pcm_subclass;
+
 	spinlock_t dpcm_lock;
 
 	bool instantiated;
@@ -1116,8 +1120,6 @@ struct snd_soc_pcm_runtime {
 	struct device *dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
-	struct mutex pcm_mutex;
-	enum snd_soc_pcm_subclass pcm_subclass;
 	struct snd_pcm_ops ops;
 
 	unsigned int params_select; /* currently selected param for dai link */
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 289211069a1e..9e54d8ae6d2c 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -80,7 +80,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream)
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) {
 		ret = cpu_dai->driver->cops->startup(cstream, cpu_dai);
@@ -108,7 +108,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream)
 
 	snd_soc_runtime_activate(rtd, cstream->direction);
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 
 	return 0;
 
@@ -118,7 +118,7 @@ machine_err:
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown)
 		cpu_dai->driver->cops->shutdown(cstream, cpu_dai);
 out:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -224,7 +224,7 @@ static void close_delayed_work(struct work_struct *work)
 			container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	dev_dbg(rtd->dev,
 		"Compress ASoC: pop wq checking: %s status: %s waiting: %s\n",
@@ -239,7 +239,7 @@ static void close_delayed_work(struct work_struct *work)
 					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 }
 
 static int soc_compr_free(struct snd_compr_stream *cstream)
@@ -249,7 +249,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	int stream;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	if (cstream->direction == SND_COMPRESS_PLAYBACK)
 		stream = SNDRV_PCM_STREAM_PLAYBACK;
@@ -292,7 +292,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
 					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return 0;
 }
 
@@ -375,7 +375,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	ret = soc_compr_components_trigger(cstream, cmd);
 	if (ret < 0)
@@ -394,7 +394,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 	}
 
 out:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -480,7 +480,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	/*
 	 * First we call set_params for the CPU DAI, then the component
@@ -514,14 +514,14 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
 
 	/* cancel any delayed stream shutdown that is pending */
 	rtd->pop_wait = 0;
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 
 	cancel_delayed_work_sync(&rtd->delayed_work);
 
 	return 0;
 
 err:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -593,7 +593,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream,
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret = 0;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_params) {
 		ret = cpu_dai->driver->cops->get_params(cstream, params, cpu_dai);
@@ -613,7 +613,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream,
 	}
 
 err:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -625,7 +625,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream,
 	struct snd_soc_rtdcom_list *rtdcom;
 	int ret = 0;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -638,7 +638,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream,
 		break;
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -650,7 +650,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream,
 	struct snd_soc_rtdcom_list *rtdcom;
 	int ret = 0;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -664,7 +664,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream,
 		break;
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -676,7 +676,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes)
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret = 0;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->ack) {
 		ret = cpu_dai->driver->cops->ack(cstream, bytes, cpu_dai);
@@ -697,7 +697,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes)
 	}
 
 err:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -710,7 +710,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream,
 	int ret = 0;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	if (cpu_dai->driver->cops && cpu_dai->driver->cops->pointer)
 		cpu_dai->driver->cops->pointer(cstream, tstamp, cpu_dai);
@@ -726,7 +726,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream,
 		break;
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -738,7 +738,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream,
 	struct snd_soc_rtdcom_list *rtdcom;
 	int ret = 0;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -751,7 +751,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream,
 		break;
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index abe2f47cee6e..b3f820fb53e6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1360,7 +1360,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 	rtd->dev->groups = soc_dev_attr_groups;
 	dev_set_name(rtd->dev, "%s", name);
 	dev_set_drvdata(rtd->dev, rtd);
-	mutex_init(&rtd->pcm_mutex);
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
@@ -2383,6 +2382,7 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	card->instantiated = 0;
 	mutex_init(&card->mutex);
 	mutex_init(&card->dapm_mutex);
+	mutex_init(&card->pcm_mutex);
 	spin_lock_init(&card->dpcm_lock);
 
 	return snd_soc_bind_card(card);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index da657c8179cc..e163dde5eab1 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -36,7 +36,7 @@
  * Increments the active count for all the DAIs and components attached to a PCM
  * runtime. Should typically be called when a stream is opened.
  *
- * Must be called with the rtd->pcm_mutex being held
+ * Must be called with the rtd->card->pcm_mutex being held
  */
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
@@ -44,7 +44,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 	struct snd_soc_dai *codec_dai;
 	int i;
 
-	lockdep_assert_held(&rtd->pcm_mutex);
+	lockdep_assert_held(&rtd->card->pcm_mutex);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		cpu_dai->playback_active++;
@@ -72,7 +72,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
  * Decrements the active count for all the DAIs and components attached to a PCM
  * runtime. Should typically be called when a stream is closed.
  *
- * Must be called with the rtd->pcm_mutex being held
+ * Must be called with the rtd->card->pcm_mutex being held
  */
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
@@ -80,7 +80,7 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
 	struct snd_soc_dai *codec_dai;
 	int i;
 
-	lockdep_assert_held(&rtd->pcm_mutex);
+	lockdep_assert_held(&rtd->card->pcm_mutex);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		cpu_dai->playback_active--;
@@ -506,7 +506,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		pm_runtime_get_sync(component->dev);
 	}
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	/* startup the audio subsystem */
 	ret = snd_soc_dai_startup(cpu_dai, substream);
@@ -604,7 +604,7 @@ dynamic:
 
 	snd_soc_runtime_activate(rtd, substream->stream);
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return 0;
 
 config_err:
@@ -623,7 +623,7 @@ component_err:
 
 	snd_soc_dai_shutdown(cpu_dai, substream);
 out:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -653,7 +653,7 @@ static void close_delayed_work(struct work_struct *work)
 			container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
 	struct snd_soc_dai *codec_dai = rtd->codec_dais[0];
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n",
 		 codec_dai->driver->playback.stream_name,
@@ -667,7 +667,7 @@ static void close_delayed_work(struct work_struct *work)
 					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 }
 
 static void codec2codec_close_delayed_work(struct work_struct *work)
@@ -694,7 +694,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai;
 	int i;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	snd_soc_runtime_deactivate(rtd, substream->stream);
 
@@ -738,7 +738,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -771,7 +771,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai;
 	int i, ret = 0;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	if (rtd->dai_link->ops->prepare) {
 		ret = rtd->dai_link->ops->prepare(substream);
@@ -826,7 +826,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);
 
 out:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -876,7 +876,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai;
 	int i, ret = 0;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 	if (rtd->dai_link->ops->hw_params) {
 		ret = rtd->dai_link->ops->hw_params(substream, params);
 		if (ret < 0) {
@@ -962,7 +962,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
         if (ret)
 		goto component_err;
 out:
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 
 component_err:
@@ -986,7 +986,7 @@ codec_err:
 	if (rtd->dai_link->ops->hw_free)
 		rtd->dai_link->ops->hw_free(substream);
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return ret;
 }
 
@@ -1001,7 +1001,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	int i;
 
-	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	/* clear the corresponding DAIs parameters when going to be inactive */
 	if (cpu_dai->active == 1) {
@@ -1043,7 +1043,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	snd_soc_dai_hw_free(cpu_dai, substream);
 
-	mutex_unlock(&rtd->pcm_mutex);
+	mutex_unlock(&rtd->card->pcm_mutex);
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From c643c189f0fec116e75f9661e734545af0e3225e Mon Sep 17 00:00:00 2001
From: Mac Chiang <mac.chiang@intel.com>
Date: Thu, 15 Aug 2019 14:51:49 +0800
Subject: ASoC: Intel: boards: Add Cometlake machine driver support

reuse and add Cometlake support with:
SSP0 for DA7219 headphone codec
SSP1 for MAX98357a speaker amp codec

Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1565851909-13825-1-git-send-email-mac.chiang@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/Kconfig                    | 27 ++++++++++++++------
 sound/soc/intel/boards/bxt_da7219_max98357a.c     | 31 ++++++++++++++++++++++-
 sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 +++++++++
 sound/soc/intel/common/soc-intel-quirks.h         |  5 ++++
 4 files changed, 66 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 4a529cea0190..5c27f7ab4a5f 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -256,16 +256,20 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH
 
 endif ## SND_SOC_INTEL_SKL
 
+config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
+	tristate
+	select SND_SOC_DA7219
+	select SND_SOC_MAX98357A
+	select SND_SOC_DMIC
+	select SND_SOC_HDAC_HDMI
+
 if SND_SOC_INTEL_APL
 
 config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
 	tristate "Broxton with DA7219 and MAX98357A in I2S Mode"
 	depends on I2C && ACPI
 	depends on MFD_INTEL_LPSS || COMPILE_TEST
-	select SND_SOC_DA7219
-	select SND_SOC_MAX98357A
-	select SND_SOC_DMIC
-	select SND_SOC_HDAC_HDMI
+	select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
 	select SND_HDA_DSP_LOADER
 	help
 	   This adds support for ASoC machine driver for Broxton-P platforms
@@ -326,10 +330,7 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
 	tristate "KBL with DA7219 and MAX98357A in I2S Mode"
 	depends on I2C && ACPI
 	depends on MFD_INTEL_LPSS || COMPILE_TEST
-	select SND_SOC_DA7219
-	select SND_SOC_MAX98357A
-	select SND_SOC_DMIC
-	select SND_SOC_HDAC_HDMI
+	select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
 	help
 	  This adds support for ASoC Onboard Codec I2S machine driver. This will
 	  create an alsa sound card for DA7219 + MAX98357A I2S audio codec.
@@ -413,4 +414,14 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
 	   If unsure select "N".
 endif ## SND_SOC_SOF_HDA_COMMON || SND_SOC_SOF_BAYTRAIL
 
+if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK)
+
+config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH
+	tristate "CML_LP with DA7219 and MAX98357A in I2S Mode"
+	depends on I2C && ACPI
+	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
+
+endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK
+
 endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index c0d865a940dc..ac1dea5f9d11 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -179,10 +179,17 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
 	int ret;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_component *component = rtd->codec_dai->component;
+	int clk_freq;
 
 	/* Configure sysclk for codec */
-	ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 19200000,
+	if (soc_intel_is_cml())
+		clk_freq = 24000000;
+	else
+		clk_freq = 19200000;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq,
 				     SND_SOC_CLOCK_IN);
+
 	if (ret) {
 		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
 		return ret;
@@ -683,6 +690,25 @@ static int broxton_audio_probe(struct platform_device *pdev)
 				broxton_dais[i].cpus->dai_name = "SSP2 Pin";
 			}
 		}
+	} else if (soc_intel_is_cml()) {
+		unsigned int i;
+
+		broxton_audio_card.name = "cmlda7219max";
+
+		for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
+			/* MAXIM_CODEC is connected to SSP1. */
+			if (!strcmp(broxton_dais[i].codecs->dai_name,
+					BXT_MAXIM_CODEC_DAI)) {
+				broxton_dais[i].name = "SSP1-Codec";
+				broxton_dais[i].cpus->dai_name = "SSP1 Pin";
+			}
+			/* DIALOG_CODEC is connected to SSP0 */
+			else if (!strcmp(broxton_dais[i].codecs->dai_name,
+					BXT_DIALOG_CODEC_DAI)) {
+				broxton_dais[i].name = "SSP0-Codec";
+				broxton_dais[i].cpus->dai_name = "SSP0 Pin";
+			}
+		}
 	}
 
 	/* override plaform name, if required */
@@ -700,6 +726,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
 static const struct platform_device_id bxt_board_ids[] = {
 	{ .name = "bxt_da7219_max98357a" },
 	{ .name = "glk_da7219_max98357a" },
+	{ .name = "cml_da7219_max98357a" },
 	{ }
 };
 
@@ -720,6 +747,8 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com>");
 MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>");
 MODULE_AUTHOR("Conrad Cooke <conrad.cooke@intel.com>");
 MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>");
+MODULE_AUTHOR("Mac Chiang <mac.chiang@intel.com>");
 MODULE_LICENSE("GPL v2");
 MODULE_ALIAS("platform:bxt_da7219_max98357a");
 MODULE_ALIAS("platform:glk_da7219_max98357a");
+MODULE_ALIAS("platform:cml_da7219_max98357a");
diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
index 771b0ef21051..985aa366c9e8 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
@@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs cml_codecs = {
 	.codecs = {"10EC5682"}
 };
 
+static struct snd_soc_acpi_codecs cml_spk_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98357A"}
+};
+
 struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = {
 	{
 		.id = "INT34C2",
@@ -28,6 +33,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = {
 		.sof_fw_filename = "sof-cnl.ri",
 		.sof_tplg_filename = "sof-cnl-rt274.tplg",
 	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "cml_da7219_max98357a",
+		.quirk_data = &cml_spk_codecs,
+		.sof_fw_filename = "sof-cnl.ri",
+		.sof_tplg_filename = "sof-cml-da7219-max98357a.tplg",
+	},
 	{
 		.id = "MX98357A",
 		.drv_name = "sof_rt5682",
diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h
index 4718fd3cf636..e6357d306cb8 100644
--- a/sound/soc/intel/common/soc-intel-quirks.h
+++ b/sound/soc/intel/common/soc-intel-quirks.h
@@ -36,6 +36,7 @@ SOC_INTEL_IS_CPU(byt, INTEL_FAM6_ATOM_SILVERMONT);
 SOC_INTEL_IS_CPU(cht, INTEL_FAM6_ATOM_AIRMONT);
 SOC_INTEL_IS_CPU(apl, INTEL_FAM6_ATOM_GOLDMONT);
 SOC_INTEL_IS_CPU(glk, INTEL_FAM6_ATOM_GOLDMONT_PLUS);
+SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE);
 
 static inline bool soc_intel_is_byt_cr(struct platform_device *pdev)
 {
@@ -110,6 +111,10 @@ static inline bool soc_intel_is_glk(void)
 	return false;
 }
 
+static inline bool soc_intel_is_cml(void)
+{
+	return false;
+}
 #endif
 
  #endif /* _SND_SOC_INTEL_QUIRKS_H */
-- 
cgit v1.2.3-59-g8ed1b


From 5f7af9ecebce7fab78b0237b0cd998f2ed3de03c Mon Sep 17 00:00:00 2001
From: Pan Xiuli <xiuli.pan@linux.intel.com>
Date: Thu, 15 Aug 2019 10:57:46 -0500
Subject: ASoC: Intel: common: add ACPI matching tables for Tiger Lake

Initial support for TGL w/ RT1308

Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190815155749.29304-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-acpi-intel-match.h              |  1 +
 sound/soc/intel/common/Makefile                   |  1 +
 sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 24 +++++++++++++++++++++++
 3 files changed, 26 insertions(+)
 create mode 100644 sound/soc/intel/common/soc-acpi-intel-tgl-match.c

(limited to 'sound')

diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h
index bb5e1e4ce8bf..ce6c4a970939 100644
--- a/include/sound/soc-acpi-intel-match.h
+++ b/include/sound/soc-acpi-intel-match.h
@@ -25,6 +25,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[];
 
 /*
  * generic table used for HDA codec-based platforms, possibly with
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 56c81e20b5bf..a14aca62ff96 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -8,6 +8,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
 	soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
 	soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
 	soc-acpi-intel-cnl-match.o soc-acpi-intel-icl-match.o \
+	soc-acpi-intel-tgl-match.o \
 	soc-acpi-intel-hda-match.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
new file mode 100644
index 000000000000..57a6298d6dca
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
@@ -0,0 +1,24 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-tgl-match.c - tables and support for ICL ACPI enumeration.
+ *
+ * Copyright (c) 2019, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = {
+	{
+		.id = "10EC1308",
+		.drv_name = "tgl_rt1308",
+		.sof_fw_filename = "sof-tgl.ri",
+		.sof_tplg_filename = "sof-tgl-rt1308.tplg",
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
-- 
cgit v1.2.3-59-g8ed1b


From 1205c81edd5ef220ca32318572e499b4b180b1f3 Mon Sep 17 00:00:00 2001
From: Pan Xiuli <xiuli.pan@linux.intel.com>
Date: Thu, 15 Aug 2019 10:57:47 -0500
Subject: ASoC: SOF: Intel: initial support for Tiger Lake.

Add Kconfig, PCI ID and chip info for Tiger Lake platform.

Note that the core mask is different from previous platforms, only
Core0 can be controlled by the host. Additional patches will be
required for multi-core functionality.

Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190815155749.29304-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/Kconfig | 16 ++++++++++++++++
 sound/soc/sof/intel/cnl.c   | 16 ++++++++++++++++
 sound/soc/sof/intel/hda.h   |  1 +
 sound/soc/sof/sof-pci-dev.c | 22 ++++++++++++++++++++++
 4 files changed, 55 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 545071afbe18..0dc5da661b61 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -27,6 +27,7 @@ config SND_SOC_SOF_INTEL_PCI
 	select SND_SOC_SOF_ICELAKE     if SND_SOC_SOF_ICELAKE_SUPPORT
 	select SND_SOC_SOF_COMETLAKE_LP if SND_SOC_SOF_COMETLAKE_LP_SUPPORT
 	select SND_SOC_SOF_COMETLAKE_H if SND_SOC_SOF_COMETLAKE_H_SUPPORT
+	select SND_SOC_SOF_TIGERLAKE   if SND_SOC_SOF_TIGERLAKE_SUPPORT
 	help
 	  This option is not user-selectable but automagically handled by
 	  'select' statements at a higher level
@@ -212,6 +213,21 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT
 	  Say Y if you have such a device.
 	  If unsure select "N".
 
+config SND_SOC_SOF_TIGERLAKE_SUPPORT
+	bool "SOF support for Tigerlake"
+	help
+          This adds support for Sound Open Firmware for Intel(R) platforms
+          using the Tigerlake processors.
+          Say Y if you have such a device.
+          If unsure select "N".
+
+config SND_SOC_SOF_TIGERLAKE
+	tristate
+	select SND_SOC_SOF_HDA_COMMON
+	help
+          This option is not user-selectable but automagically handled by
+	  'select' statements at a higher level
+
 config SND_SOC_SOF_HDA_COMMON
 	tristate
 	select SND_SOC_SOF_INTEL_COMMON
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index 6d7d9c93252c..5de281fcc122 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -295,3 +295,19 @@ const struct sof_intel_dsp_desc icl_chip_info = {
 	.ssp_base_offset = CNL_SSP_BASE_OFFSET,
 };
 EXPORT_SYMBOL(icl_chip_info);
+
+const struct sof_intel_dsp_desc tgl_chip_info = {
+	/* Tigerlake */
+	.cores_num = 4,
+	.init_core_mask = 1,
+	.cores_mask = HDA_DSP_CORE_MASK(0),
+	.ipc_req = CNL_DSP_REG_HIPCIDR,
+	.ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY,
+	.ipc_ack = CNL_DSP_REG_HIPCIDA,
+	.ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE,
+	.ipc_ctl = CNL_DSP_REG_HIPCCTL,
+	.rom_init_timeout	= 300,
+	.ssp_count = ICL_SSP_COUNT,
+	.ssp_base_offset = CNL_SSP_BASE_OFFSET,
+};
+EXPORT_SYMBOL(tgl_chip_info);
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 6db3dcbab3bd..cbb431f7835d 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -599,5 +599,6 @@ extern const struct sof_intel_dsp_desc apl_chip_info;
 extern const struct sof_intel_dsp_desc cnl_chip_info;
 extern const struct sof_intel_dsp_desc skl_chip_info;
 extern const struct sof_intel_dsp_desc icl_chip_info;
+extern const struct sof_intel_dsp_desc tgl_chip_info;
 
 #endif
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index 6fd3df7c57a3..9f92504ad5b4 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -203,6 +203,24 @@ static const struct sof_dev_desc kbl_desc = {
 };
 #endif
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
+static const struct sof_dev_desc tgl_desc = {
+	.machines               = snd_soc_acpi_intel_tgl_machines,
+	.resindex_lpe_base      = 0,
+	.resindex_pcicfg_base   = -1,
+	.resindex_imr_base      = -1,
+	.irqindex_host_ipc      = -1,
+	.resindex_dma_base      = -1,
+	.chip_info = &tgl_chip_info,
+	.default_fw_path = "intel/sof",
+	.default_tplg_path = "intel/sof-tplg",
+	.nocodec_fw_filename = "sof-tgl.ri",
+	.nocodec_tplg_filename = "sof-tgl-nocodec.tplg",
+	.ops = &sof_cnl_ops,
+	.arch_ops = &sof_xtensa_arch_ops
+};
+#endif
+
 static const struct dev_pm_ops sof_pci_pm = {
 	SET_SYSTEM_SLEEP_PM_OPS(snd_sof_suspend, snd_sof_resume)
 	SET_RUNTIME_PM_OPS(snd_sof_runtime_suspend, snd_sof_runtime_resume,
@@ -384,6 +402,10 @@ static const struct pci_device_id sof_pci_ids[] = {
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
 	{ PCI_DEVICE(0x8086, 0x06c8),
 		.driver_data = (unsigned long)&cml_desc},
+#endif
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
+	{ PCI_DEVICE(0x8086, 0xa0c8),
+		.driver_data = (unsigned long)&tgl_desc},
 #endif
 	{ 0, }
 };
-- 
cgit v1.2.3-59-g8ed1b


From f4ead53241c7dcd2f3130e059cb3dfdec0ec9871 Mon Sep 17 00:00:00 2001
From: Pan Xiuli <xiuli.pan@linux.intel.com>
Date: Thu, 15 Aug 2019 10:57:48 -0500
Subject: ASoC: Intel: common: add ACPI matching tables for EHL

There are no upstream machine drivers just yet so just add dummy table
for compilation in nocodec-mode.

Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190815155749.29304-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-acpi-intel-match.h              |  1 +
 sound/soc/intel/common/Makefile                   |  2 +-
 sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 18 ++++++++++++++++++
 3 files changed, 20 insertions(+), 1 deletion(-)
 create mode 100644 sound/soc/intel/common/soc-acpi-intel-ehl-match.c

(limited to 'sound')

diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h
index ce6c4a970939..6c9929abd90b 100644
--- a/include/sound/soc-acpi-intel-match.h
+++ b/include/sound/soc-acpi-intel-match.h
@@ -26,6 +26,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[];
 
 /*
  * generic table used for HDA codec-based platforms, possibly with
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index a14aca62ff96..18d9630ae9a2 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -8,7 +8,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
 	soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
 	soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
 	soc-acpi-intel-cnl-match.o soc-acpi-intel-icl-match.o \
-	soc-acpi-intel-tgl-match.o \
+	soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \
 	soc-acpi-intel-hda-match.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
new file mode 100644
index 000000000000..a1290c3fa99f
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
@@ -0,0 +1,18 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * soc-apci-intel-ehl-match.c - tables and support for EHL ACPI enumeration.
+ *
+ * Copyright (c) 2019, Intel Corporation.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[] = {
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ehl_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
-- 
cgit v1.2.3-59-g8ed1b


From 6173269038e98408c27cfb438ff07d463fbd0b33 Mon Sep 17 00:00:00 2001
From: Pan Xiuli <xiuli.pan@linux.intel.com>
Date: Thu, 15 Aug 2019 10:57:49 -0500
Subject: ASoC: SOF: Intel: initial support for Elkhart Lake

Add Kconfig, PCI ID and chip info for EHL platform.

Note that the core mask is different from previous platforms, only
Core0 can be controlled by the host. Additional patches will be
required for multi-core functionality.

Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190815155749.29304-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/Kconfig | 16 ++++++++++++++++
 sound/soc/sof/intel/cnl.c   | 16 ++++++++++++++++
 sound/soc/sof/intel/hda.h   |  1 +
 sound/soc/sof/sof-pci-dev.c | 22 ++++++++++++++++++++++
 4 files changed, 55 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 0dc5da661b61..889b6202d054 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -28,6 +28,7 @@ config SND_SOC_SOF_INTEL_PCI
 	select SND_SOC_SOF_COMETLAKE_LP if SND_SOC_SOF_COMETLAKE_LP_SUPPORT
 	select SND_SOC_SOF_COMETLAKE_H if SND_SOC_SOF_COMETLAKE_H_SUPPORT
 	select SND_SOC_SOF_TIGERLAKE   if SND_SOC_SOF_TIGERLAKE_SUPPORT
+	select SND_SOC_SOF_ELKHARTLAKE if SND_SOC_SOF_ELKHARTLAKE_SUPPORT
 	help
 	  This option is not user-selectable but automagically handled by
 	  'select' statements at a higher level
@@ -228,6 +229,21 @@ config SND_SOC_SOF_TIGERLAKE
           This option is not user-selectable but automagically handled by
 	  'select' statements at a higher level
 
+config SND_SOC_SOF_ELKHARTLAKE_SUPPORT
+	bool "SOF support for ElkhartLake"
+	help
+          This adds support for Sound Open Firmware for Intel(R) platforms
+          using the ElkhartLake processors.
+          Say Y if you have such a device.
+          If unsure select "N".
+
+config SND_SOC_SOF_ELKHARTLAKE
+	tristate
+	select SND_SOC_SOF_HDA_COMMON
+	help
+          This option is not user-selectable but automagically handled by
+	  'select' statements at a higher level
+
 config SND_SOC_SOF_HDA_COMMON
 	tristate
 	select SND_SOC_SOF_INTEL_COMMON
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index 5de281fcc122..4ddd73762d81 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -311,3 +311,19 @@ const struct sof_intel_dsp_desc tgl_chip_info = {
 	.ssp_base_offset = CNL_SSP_BASE_OFFSET,
 };
 EXPORT_SYMBOL(tgl_chip_info);
+
+const struct sof_intel_dsp_desc ehl_chip_info = {
+	/* Elkhartlake */
+	.cores_num = 4,
+	.init_core_mask = 1,
+	.cores_mask = HDA_DSP_CORE_MASK(0),
+	.ipc_req = CNL_DSP_REG_HIPCIDR,
+	.ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY,
+	.ipc_ack = CNL_DSP_REG_HIPCIDA,
+	.ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE,
+	.ipc_ctl = CNL_DSP_REG_HIPCCTL,
+	.rom_init_timeout	= 300,
+	.ssp_count = ICL_SSP_COUNT,
+	.ssp_base_offset = CNL_SSP_BASE_OFFSET,
+};
+EXPORT_SYMBOL(ehl_chip_info);
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index cbb431f7835d..5591841a1b6f 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -600,5 +600,6 @@ extern const struct sof_intel_dsp_desc cnl_chip_info;
 extern const struct sof_intel_dsp_desc skl_chip_info;
 extern const struct sof_intel_dsp_desc icl_chip_info;
 extern const struct sof_intel_dsp_desc tgl_chip_info;
+extern const struct sof_intel_dsp_desc ehl_chip_info;
 
 #endif
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index 9f92504ad5b4..d66412a77873 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -221,6 +221,24 @@ static const struct sof_dev_desc tgl_desc = {
 };
 #endif
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE)
+static const struct sof_dev_desc ehl_desc = {
+	.machines               = snd_soc_acpi_intel_ehl_machines,
+	.resindex_lpe_base      = 0,
+	.resindex_pcicfg_base   = -1,
+	.resindex_imr_base      = -1,
+	.irqindex_host_ipc      = -1,
+	.resindex_dma_base      = -1,
+	.chip_info = &ehl_chip_info,
+	.default_fw_path = "intel/sof",
+	.default_tplg_path = "intel/sof-tplg",
+	.nocodec_fw_filename = "sof-ehl.ri",
+	.nocodec_tplg_filename = "sof-ehl-nocodec.tplg",
+	.ops = &sof_cnl_ops,
+	.arch_ops = &sof_xtensa_arch_ops
+};
+#endif
+
 static const struct dev_pm_ops sof_pci_pm = {
 	SET_SYSTEM_SLEEP_PM_OPS(snd_sof_suspend, snd_sof_resume)
 	SET_RUNTIME_PM_OPS(snd_sof_runtime_suspend, snd_sof_runtime_resume,
@@ -406,6 +424,10 @@ static const struct pci_device_id sof_pci_ids[] = {
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
 	{ PCI_DEVICE(0x8086, 0xa0c8),
 		.driver_data = (unsigned long)&tgl_desc},
+#endif
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE)
+	{ PCI_DEVICE(0x8086, 0x4b55),
+		.driver_data = (unsigned long)&ehl_desc},
 #endif
 	{ 0, }
 };
-- 
cgit v1.2.3-59-g8ed1b


From 246bb4aaa4f4314816ae6835f3c6cc6dc7a10f4f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 15 Aug 2019 21:42:51 +0200
Subject: Revert "ALSA: hda: Add codec on bus address table lately"

This reverts commit ee5f85d9290f ("ALSA: hda: Add codec on bus address
table lately").  The commit caused several regression since I've
overlooked that the function doesn't manage only the caddr_tbl but
also the codec linked list that is referred indirectly in the other
drivers.

Revert for now to make everything back to work.

Fixes: ee5f85d9290f ("ALSA: hda: Add codec on bus address table lately")
Reported-by: Chris Wilson <chris@chris-wilson.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_device.c | 21 +++++++++------------
 1 file changed, 9 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index bf83d7062ef6..9f3e37511408 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -61,6 +61,10 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus,
 	pm_runtime_get_noresume(&codec->dev);
 	atomic_set(&codec->in_pm, 0);
 
+	err = snd_hdac_bus_add_device(bus, codec);
+	if (err < 0)
+		goto error;
+
 	/* fill parameters */
 	codec->vendor_id = snd_hdac_read_parm(codec, AC_NODE_ROOT,
 					      AC_PAR_VENDOR_ID);
@@ -139,22 +143,15 @@ int snd_hdac_device_register(struct hdac_device *codec)
 	err = device_add(&codec->dev);
 	if (err < 0)
 		return err;
-	err = snd_hdac_bus_add_device(codec->bus, codec);
-	if (err < 0)
-		goto error;
 	mutex_lock(&codec->widget_lock);
 	err = hda_widget_sysfs_init(codec);
 	mutex_unlock(&codec->widget_lock);
-	if (err < 0)
-		goto error_remove;
+	if (err < 0) {
+		device_del(&codec->dev);
+		return err;
+	}
 
 	return 0;
-
- error_remove:
-	snd_hdac_bus_remove_device(codec->bus, codec);
- error:
-	device_del(&codec->dev);
-	return err;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_device_register);
 
@@ -168,8 +165,8 @@ void snd_hdac_device_unregister(struct hdac_device *codec)
 		mutex_lock(&codec->widget_lock);
 		hda_widget_sysfs_exit(codec);
 		mutex_unlock(&codec->widget_lock);
-		snd_hdac_bus_remove_device(codec->bus, codec);
 		device_del(&codec->dev);
+		snd_hdac_bus_remove_device(codec->bus, codec);
 	}
 }
 EXPORT_SYMBOL_GPL(snd_hdac_device_unregister);
-- 
cgit v1.2.3-59-g8ed1b


From 0fc1e447e9e474c2460df8d0378f899b8813e1d2 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Fri, 16 Aug 2019 14:27:39 +0800
Subject: ALSA: hda - Expand pin_match function to match upcoming new tbls

With the existing pintbl, we already have many entries in it. it is
better to figure out a new way to reduce the size of the pintbl.

We plan to define a new tbl which will match more machines with a
single tbl, To do that, this function doesn't need to match all valid
pins between machine and tbl, it just needs to match all pins defined
in the tbl with the machine.

And the plan is to move some tbls from pin_fixup_tbl to
fallback_pin_fixup_tbl gradually.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c | 12 ++++++++----
 sound/pci/hda/hda_local.h       |  3 ++-
 sound/pci/hda/patch_realtek.c   |  4 ++--
 3 files changed, 12 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 92390d457567..8a134f750bae 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -884,7 +884,8 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup);
 #define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC))
 
 static bool pin_config_match(struct hda_codec *codec,
-			     const struct hda_pintbl *pins)
+			     const struct hda_pintbl *pins,
+			     bool match_all_pins)
 {
 	const struct hda_pincfg *pin;
 	int i;
@@ -908,7 +909,8 @@ static bool pin_config_match(struct hda_codec *codec,
 					return false;
 			}
 		}
-		if (!found && (cfg & 0xf0000000) != 0x40000000)
+		if (match_all_pins &&
+		    !found && (cfg & 0xf0000000) != 0x40000000)
 			return false;
 	}
 
@@ -920,10 +922,12 @@ static bool pin_config_match(struct hda_codec *codec,
  * @codec: the HDA codec
  * @pin_quirk: zero-terminated pin quirk list
  * @fixlist: the fixup list
+ * @match_all_pins: all valid pins must match with the table entries
  */
 void snd_hda_pick_pin_fixup(struct hda_codec *codec,
 			    const struct snd_hda_pin_quirk *pin_quirk,
-			    const struct hda_fixup *fixlist)
+			    const struct hda_fixup *fixlist,
+			    bool match_all_pins)
 {
 	const struct snd_hda_pin_quirk *pq;
 
@@ -935,7 +939,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec,
 			continue;
 		if (codec->core.vendor_id != pq->codec)
 			continue;
-		if (pin_config_match(codec, pq->pins)) {
+		if (pin_config_match(codec, pq->pins, match_all_pins)) {
 			codec->fixup_id = pq->value;
 #ifdef CONFIG_SND_DEBUG_VERBOSE
 			codec->fixup_name = pq->name;
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 349a8312d06a..3942e1b528d8 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -361,7 +361,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 			const struct hda_fixup *fixlist);
 void snd_hda_pick_pin_fixup(struct hda_codec *codec,
 			    const struct snd_hda_pin_quirk *pin_quirk,
-			    const struct hda_fixup *fixlist);
+			    const struct hda_fixup *fixlist,
+			    bool match_all_pins);
 
 /* helper macros to retrieve pin default-config values */
 #define get_defcfg_connect(cfg) \
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cb7baa65b298..9562045c49c6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7958,7 +7958,7 @@ static int patch_alc269(struct hda_codec *codec)
 
 	snd_hda_pick_fixup(codec, alc269_fixup_models,
 		       alc269_fixup_tbl, alc269_fixups);
-	snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups);
+	snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true);
 	snd_hda_pick_fixup(codec, NULL,	alc269_fixup_vendor_tbl,
 			   alc269_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -8943,7 +8943,7 @@ static int patch_alc662(struct hda_codec *codec)
 
 	snd_hda_pick_fixup(codec, alc662_fixup_models,
 		       alc662_fixup_tbl, alc662_fixups);
-	snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups);
+	snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups, true);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
-- 
cgit v1.2.3-59-g8ed1b


From 7c0a69394c265f2bb674c3f5daadfdd5c15ea0d1 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Fri, 16 Aug 2019 14:27:40 +0800
Subject: ALSA: hda - Define a fallback_pin_fixup_tbl for alc269 family

We have another Dell laptop which needs the DELL4_MIC_NO_PRESENCE,
and this laptop has different pincfg definitions from existing
ones in the pintbl, rather adding a new entry, let us define
a tbl in the fallback_pin_fixup_tbl and this tbl will match
all dell machines with alc289 codec and the pins of 0x19 and 0x1b
are undef by default.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 18 ++++++++++++++----
 1 file changed, 14 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9562045c49c6..b6ae452a1d8f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7656,10 +7656,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x12, 0x90a60120},
 		{0x14, 0x90170110},
 		{0x21, 0x0321101f}),
-	SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
-		{0x12, 0xb7a60130},
-		{0x14, 0x90170110},
-		{0x21, 0x04211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1,
 		ALC290_STANDARD_PINS,
 		{0x15, 0x04211040},
@@ -7769,6 +7765,19 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 	{}
 };
 
+/* This is the fallback pin_fixup_tbl for alc269 family, to make the tbl match
+ * more machines, don't need to match all valid pins, just need to match
+ * all the pins defined in the tbl. Just because of this reason, it is possible
+ * that a single machine matches multiple tbls, so there is one limitation:
+ *   at most one tbl is allowed to define for the same vendor and same codec
+ */
+static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = {
+	SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
+		{0x19, 0x40000000},
+		{0x1b, 0x40000000}),
+	{}
+};
+
 static void alc269_fill_coef(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -7959,6 +7968,7 @@ static int patch_alc269(struct hda_codec *codec)
 	snd_hda_pick_fixup(codec, alc269_fixup_models,
 		       alc269_fixup_tbl, alc269_fixups);
 	snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true);
+	snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false);
 	snd_hda_pick_fixup(codec, NULL,	alc269_fixup_vendor_tbl,
 			   alc269_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
-- 
cgit v1.2.3-59-g8ed1b


From 8661ab5b23d6d30d8687fc05bc1dba8f9a64b444 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Fri, 16 Aug 2019 01:03:14 -0400
Subject: ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power
source of modules during system suspend and resume procedure.
Thus, AUDMUX needs to save all the values of registers before the
system suspend and restore them after the system resume.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1565931794-7218-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/imx-audmux.c | 54 +++++++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 53 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index b2351cd33b0f..16ede3b5cb32 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -23,6 +23,8 @@
 
 static struct clk *audmux_clk;
 static void __iomem *audmux_base;
+static u32 *regcache;
+static u32 reg_max;
 
 #define IMX_AUDMUX_V2_PTCR(x)		((x) * 8)
 #define IMX_AUDMUX_V2_PDCR(x)		((x) * 8 + 4)
@@ -317,8 +319,23 @@ static int imx_audmux_probe(struct platform_device *pdev)
 	if (of_id)
 		pdev->id_entry = of_id->data;
 	audmux_type = pdev->id_entry->driver_data;
-	if (audmux_type == IMX31_AUDMUX)
+
+	switch (audmux_type) {
+	case IMX31_AUDMUX:
 		audmux_debugfs_init();
+		reg_max = 14;
+		break;
+	case IMX21_AUDMUX:
+		reg_max = 6;
+		break;
+	default:
+		dev_err(&pdev->dev, "unsupported version!\n");
+		return -EINVAL;
+	}
+
+	regcache = devm_kzalloc(&pdev->dev, sizeof(u32) * reg_max, GFP_KERNEL);
+	if (!regcache)
+		return -ENOMEM;
 
 	if (of_id)
 		imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
@@ -334,12 +351,47 @@ static int imx_audmux_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int imx_audmux_suspend(struct device *dev)
+{
+	int i;
+
+	clk_prepare_enable(audmux_clk);
+
+	for (i = 0; i < reg_max; i++)
+		regcache[i] = readl(audmux_base + i * 4);
+
+	clk_disable_unprepare(audmux_clk);
+
+	return 0;
+}
+
+static int imx_audmux_resume(struct device *dev)
+{
+	int i;
+
+	clk_prepare_enable(audmux_clk);
+
+	for (i = 0; i < reg_max; i++)
+		writel(regcache[i], audmux_base + i * 4);
+
+	clk_disable_unprepare(audmux_clk);
+
+	return 0;
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops imx_audmux_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(imx_audmux_suspend, imx_audmux_resume)
+};
+
 static struct platform_driver imx_audmux_driver = {
 	.probe		= imx_audmux_probe,
 	.remove		= imx_audmux_remove,
 	.id_table	= imx_audmux_ids,
 	.driver	= {
 		.name	= DRIVER_NAME,
+		.pm = &imx_audmux_pm,
 		.of_match_table = imx_audmux_dt_ids,
 	}
 };
-- 
cgit v1.2.3-59-g8ed1b


From f59b16ef4ccea1b52f1c4e4c60ce507dc0bcc0ad Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Thu, 15 Aug 2019 14:20:15 -0500
Subject: ASoC: SOF: topology: Add dummy support for i.MX8 DAIs

Add dummy support for SAI/ESAI digital audio interface
IPs found on i.MX8 boards.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190815192018.30570-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/sof/dai.h         |  2 ++
 include/uapi/sound/sof/tokens.h |  8 ++++++++
 sound/soc/sof/topology.c        | 30 ++++++++++++++++++++++++++++++
 3 files changed, 40 insertions(+)

(limited to 'sound')

diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h
index da9825ad41d4..86494294274e 100644
--- a/include/sound/sof/dai.h
+++ b/include/sound/sof/dai.h
@@ -50,6 +50,8 @@ enum sof_ipc_dai_type {
 	SOF_DAI_INTEL_DMIC,		/**< Intel DMIC */
 	SOF_DAI_INTEL_HDA,		/**< Intel HD/A */
 	SOF_DAI_INTEL_SOUNDWIRE,	/**< Intel SoundWire */
+	SOF_DAI_IMX_SAI,		/**< i.MX SAI */
+	SOF_DAI_IMX_ESAI,		/**< i.MX ESAI */
 };
 
 /* general purpose DAI configuration */
diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h
index 6435240cef13..8f996857fb24 100644
--- a/include/uapi/sound/sof/tokens.h
+++ b/include/uapi/sound/sof/tokens.h
@@ -106,4 +106,12 @@
 /* for backward compatibility */
 #define SOF_TKN_EFFECT_TYPE	SOF_TKN_PROCESS_TYPE
 
+/* SAI */
+#define SOF_TKN_IMX_SAI_FIRST_TOKEN		1000
+/* TODO: Add SAI tokens */
+
+/* ESAI */
+#define SOF_TKN_IMX_ESAI_FIRST_TOKEN		1100
+/* TODO: Add ESAI tokens */
+
 #endif
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 9cffea142395..a215bf58b138 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -346,6 +346,8 @@ static const struct sof_dai_types sof_dais[] = {
 	{"SSP", SOF_DAI_INTEL_SSP},
 	{"HDA", SOF_DAI_INTEL_HDA},
 	{"DMIC", SOF_DAI_INTEL_DMIC},
+	{"SAI", SOF_DAI_IMX_SAI},
+	{"ESAI", SOF_DAI_IMX_ESAI},
 };
 
 static enum sof_ipc_dai_type find_dai(const char *name)
@@ -2513,6 +2515,26 @@ static int sof_link_ssp_load(struct snd_soc_component *scomp, int index,
 	return ret;
 }
 
+static int sof_link_sai_load(struct snd_soc_component *scomp, int index,
+			     struct snd_soc_dai_link *link,
+			     struct snd_soc_tplg_link_config *cfg,
+			     struct snd_soc_tplg_hw_config *hw_config,
+			     struct sof_ipc_dai_config *config)
+{
+	/*TODO: Add implementation */
+	return 0;
+}
+
+static int sof_link_esai_load(struct snd_soc_component *scomp, int index,
+			      struct snd_soc_dai_link *link,
+			      struct snd_soc_tplg_link_config *cfg,
+			      struct snd_soc_tplg_hw_config *hw_config,
+			      struct sof_ipc_dai_config *config)
+{
+	/*TODO: Add implementation */
+	return 0;
+}
+
 static int sof_link_dmic_load(struct snd_soc_component *scomp, int index,
 			      struct snd_soc_dai_link *link,
 			      struct snd_soc_tplg_link_config *cfg,
@@ -2837,6 +2859,14 @@ static int sof_link_load(struct snd_soc_component *scomp, int index,
 		ret = sof_link_hda_load(scomp, index, link, cfg, hw_config,
 					&config);
 		break;
+	case SOF_DAI_IMX_SAI:
+		ret = sof_link_sai_load(scomp, index, link, cfg, hw_config,
+					&config);
+		break;
+	case SOF_DAI_IMX_ESAI:
+		ret = sof_link_esai_load(scomp, index, link, cfg, hw_config,
+					 &config);
+		break;
 	default:
 		dev_err(sdev->dev, "error: invalid DAI type %d\n", config.type);
 		ret = -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From 4d6bbf1a015c84ba05af9d8c59dd9c55cad4e668 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 15 Aug 2019 14:20:17 -0500
Subject: ASoC: SOF: topology: initial support for Intel ALH DAI type

The Audio Link Hub DAI does not require any static configuration from
topology for now. We still need to pass the frame rate and format to
firmware.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190815192018.30570-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/topology.c | 42 +++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 41 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index a215bf58b138..28a7a6e06a53 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -346,6 +346,7 @@ static const struct sof_dai_types sof_dais[] = {
 	{"SSP", SOF_DAI_INTEL_SSP},
 	{"HDA", SOF_DAI_INTEL_HDA},
 	{"DMIC", SOF_DAI_INTEL_DMIC},
+	{"ALH", SOF_DAI_INTEL_ALH},
 	{"SAI", SOF_DAI_IMX_SAI},
 	{"ESAI", SOF_DAI_IMX_ESAI},
 };
@@ -2763,6 +2764,40 @@ static int sof_link_hda_load(struct snd_soc_component *scomp, int index,
 	return ret;
 }
 
+static int sof_link_alh_load(struct snd_soc_component *scomp, int index,
+			     struct snd_soc_dai_link *link,
+			     struct snd_soc_tplg_link_config *cfg,
+			     struct snd_soc_tplg_hw_config *hw_config,
+			     struct sof_ipc_dai_config *config)
+{
+	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
+	struct sof_ipc_reply reply;
+	u32 size = sizeof(*config);
+	int ret;
+
+	/* init IPC */
+	config->hdr.size = size;
+
+	/* send message to DSP */
+	ret = sof_ipc_tx_message(sdev->ipc,
+				 config->hdr.cmd, config, size, &reply,
+				 sizeof(reply));
+
+	if (ret < 0) {
+		dev_err(sdev->dev, "error: failed to set DAI config for ALH %d\n",
+			config->dai_index);
+		return ret;
+	}
+
+	/* set config for all DAI's with name matching the link name */
+	ret = sof_set_dai_config(sdev, size, link, config);
+	if (ret < 0)
+		dev_err(sdev->dev, "error: failed to save DAI config for ALH %d\n",
+			config->dai_index);
+
+	return ret;
+}
+
 /* DAI link - used for any driver specific init */
 static int sof_link_load(struct snd_soc_component *scomp, int index,
 			 struct snd_soc_dai_link *link,
@@ -2859,6 +2894,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index,
 		ret = sof_link_hda_load(scomp, index, link, cfg, hw_config,
 					&config);
 		break;
+	case SOF_DAI_INTEL_ALH:
+		ret = sof_link_alh_load(scomp, index, link, cfg, hw_config,
+					&config);
+		break;
 	case SOF_DAI_IMX_SAI:
 		ret = sof_link_sai_load(scomp, index, link, cfg, hw_config,
 					&config);
@@ -2924,7 +2963,8 @@ found:
 	switch (sof_dai->dai_config->type) {
 	case SOF_DAI_INTEL_SSP:
 	case SOF_DAI_INTEL_DMIC:
-		/* no resource needs to be released for SSP and DMIC */
+	case SOF_DAI_INTEL_ALH:
+		/* no resource needs to be released for SSP, DMIC and ALH */
 		break;
 	case SOF_DAI_INTEL_HDA:
 		ret = sof_link_hda_unload(sdev, link);
-- 
cgit v1.2.3-59-g8ed1b


From 6e3360cd07ec2d4a53919dc62947f9769d8fdac4 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 15 Aug 2019 14:20:18 -0500
Subject: ASoC: SOF: pcm: add ALH support

Even if ALH has no specific configuration, we still need to handle the
common parameters for all DAIs

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190815192018.30570-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/pcm.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 8612896673a5..e3f6a6dc0f36 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -675,6 +675,9 @@ static int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
 	case SOF_DAI_INTEL_HDA:
 		/* do nothing for HDA dai_link */
 		break;
+	case SOF_DAI_INTEL_ALH:
+		/* do nothing for ALH dai_link */
+		break;
 	default:
 		dev_err(sdev->dev, "error: invalid DAI type %d\n",
 			dai->dai_config->type);
-- 
cgit v1.2.3-59-g8ed1b


From c372a35550c8d60f673b20210eea58a06d6d38cb Mon Sep 17 00:00:00 2001
From: Kunihiko Hayashi <hayashi.kunihiko@socionext.com>
Date: Tue, 20 Aug 2019 15:16:04 +0900
Subject: ASoC: uniphier: Fix double reset assersion when transitioning to
 suspend state

When transitioning to supend state, uniphier_aio_dai_suspend() is called
and asserts reset lines and disables clocks.

However, if there are two or more DAIs, uniphier_aio_dai_suspend() are
called multiple times, and double reset assersion will cause.

This patch defines the counter that has the number of DAIs at first, and
whenever uniphier_aio_dai_suspend() are called, it decrements the
counter. And only if the counter is zero, it asserts reset lines and
disables clocks.

In the same way, uniphier_aio_dai_resume() are called, it increments the
counter after deasserting reset lines and enabling clocks.

Fixes: 139a34200233 ("ASoC: uniphier: add support for UniPhier AIO CPU DAI driver")
Signed-off-by: Kunihiko Hayashi <hayashi.kunihiko@socionext.com>
Link: https://lore.kernel.org/r/1566281764-14059-1-git-send-email-hayashi.kunihiko@socionext.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/uniphier/aio-cpu.c | 31 +++++++++++++++++++++----------
 sound/soc/uniphier/aio.h     |  1 +
 2 files changed, 22 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c
index ee90e6c3937c..2ae582a99b63 100644
--- a/sound/soc/uniphier/aio-cpu.c
+++ b/sound/soc/uniphier/aio-cpu.c
@@ -424,8 +424,11 @@ int uniphier_aio_dai_suspend(struct snd_soc_dai *dai)
 {
 	struct uniphier_aio *aio = uniphier_priv(dai);
 
-	reset_control_assert(aio->chip->rst);
-	clk_disable_unprepare(aio->chip->clk);
+	aio->chip->num_wup_aios--;
+	if (!aio->chip->num_wup_aios) {
+		reset_control_assert(aio->chip->rst);
+		clk_disable_unprepare(aio->chip->clk);
+	}
 
 	return 0;
 }
@@ -439,13 +442,15 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai)
 	if (!aio->chip->active)
 		return 0;
 
-	ret = clk_prepare_enable(aio->chip->clk);
-	if (ret)
-		return ret;
+	if (!aio->chip->num_wup_aios) {
+		ret = clk_prepare_enable(aio->chip->clk);
+		if (ret)
+			return ret;
 
-	ret = reset_control_deassert(aio->chip->rst);
-	if (ret)
-		goto err_out_clock;
+		ret = reset_control_deassert(aio->chip->rst);
+		if (ret)
+			goto err_out_clock;
+	}
 
 	aio_iecout_set_enable(aio->chip, true);
 	aio_chip_init(aio->chip);
@@ -458,7 +463,7 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai)
 
 		ret = aio_init(sub);
 		if (ret)
-			goto err_out_clock;
+			goto err_out_reset;
 
 		if (!sub->setting)
 			continue;
@@ -466,11 +471,16 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai)
 		aio_port_reset(sub);
 		aio_src_reset(sub);
 	}
+	aio->chip->num_wup_aios++;
 
 	return 0;
 
+err_out_reset:
+	if (!aio->chip->num_wup_aios)
+		reset_control_assert(aio->chip->rst);
 err_out_clock:
-	clk_disable_unprepare(aio->chip->clk);
+	if (!aio->chip->num_wup_aios)
+		clk_disable_unprepare(aio->chip->clk);
 
 	return ret;
 }
@@ -619,6 +629,7 @@ int uniphier_aio_probe(struct platform_device *pdev)
 		return PTR_ERR(chip->rst);
 
 	chip->num_aios = chip->chip_spec->num_dais;
+	chip->num_wup_aios = chip->num_aios;
 	chip->aios = devm_kcalloc(dev,
 				  chip->num_aios, sizeof(struct uniphier_aio),
 				  GFP_KERNEL);
diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h
index ca6ccbae0ee8..a7ff7e556429 100644
--- a/sound/soc/uniphier/aio.h
+++ b/sound/soc/uniphier/aio.h
@@ -285,6 +285,7 @@ struct uniphier_aio_chip {
 
 	struct uniphier_aio *aios;
 	int num_aios;
+	int num_wup_aios;
 	struct uniphier_aio_pll *plls;
 	int num_plls;
 
-- 
cgit v1.2.3-59-g8ed1b


From 18dd62ae3bc31baa0473e4a09e46c02e0bdc57a0 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Tue, 20 Aug 2019 14:34:13 +0200
Subject: ASoC: meson: axg-tdm-formatter: free reset on device removal

Use the devm variant to get the formatter reset so it is properly freed
on device removal

Fixes: 751bd5db5260 ("ASoC: meson: axg-tdm-formatter: add reset")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190820123413.22249-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-tdm-formatter.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 2e498201139f..1a0bf9d3836d 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -327,7 +327,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev)
 	}
 
 	/* Formatter dedicated reset line */
-	formatter->reset = reset_control_get_optional_exclusive(dev, NULL);
+	formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL);
 	if (IS_ERR(formatter->reset)) {
 		ret = PTR_ERR(formatter->reset);
 		if (ret != -EPROBE_DEFER)
-- 
cgit v1.2.3-59-g8ed1b


From 351b31002c1853af078ebfffd4b67bfc3d19e3dd Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Tue, 20 Aug 2019 14:35:10 +0200
Subject: ASoC: meson: g12a-tohdmitx: require regmap mmio

The tohdmitx glue uses regmap MMIO so it should require it.

Fixes: c8609f3870f7 ("ASoC: meson: add g12a tohdmitx control")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190820123510.22491-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/Kconfig | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 63b38c123103..2e3676147cea 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -87,6 +87,7 @@ config SND_MESON_AXG_PDM
 
 config SND_MESON_G12A_TOHDMITX
 	tristate "Amlogic G12A To HDMI TX Control Support"
+	select REGMAP_MMIO
 	imply SND_SOC_HDMI_CODEC
 	help
 	  Select Y or M to add support for HDMI audio on the g12a SoC
-- 
cgit v1.2.3-59-g8ed1b


From 404be07f4ed27697f5fa69162f67a94555738595 Mon Sep 17 00:00:00 2001
From: Daniel Stuart <daniel.stuart14@gmail.com>
Date: Thu, 15 Aug 2019 14:12:55 -0300
Subject: ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need
 pmc_plt_clk_0 quirk

Every single baytrail chromebook sets PMC to 0, as can be seeing
below by searching through coreboot source code:
	$ grep -rl "PMC_PLT_CLK\[0\]" .
	./rambi/variants/glimmer/devicetree.cb
	./rambi/variants/clapper/devicetree.cb
	./rambi/variants/swanky/devicetree.cb
	./rambi/variants/enguarde/devicetree.cb
	./rambi/variants/winky/devicetree.cb
	./rambi/variants/kip/devicetree.cb
	./rambi/variants/squawks/devicetree.cb
	./rambi/variants/orco/devicetree.cb
	./rambi/variants/ninja/devicetree.cb
	./rambi/variants/heli/devicetree.cb
	./rambi/variants/sumo/devicetree.cb
	./rambi/variants/banjo/devicetree.cb
	./rambi/variants/candy/devicetree.cb
	./rambi/variants/gnawty/devicetree.cb
	./rambi/variants/rambi/devicetree.cb
	./rambi/variants/quawks/devicetree.cb

Plus, Cyan (only non-baytrail chromebook with max98090) also needs
this patch for audio to work.

Thus, this commit adds all the missing devices to bsw_max98090 quirk
table, implemented by commit a182ecd3809c ("ASoC: intel:
cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0").

Signed-off-by: Daniel Stuart <daniel.stuart14@gmail.com>
Link: https://lore.kernel.org/r/20190815171300.30126-1-daniel.stuart14@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++++++++++
 1 file changed, 98 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 33eb72545be6..83b978e7b4c4 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -399,6 +399,20 @@ static struct snd_soc_card snd_soc_card_cht = {
 };
 
 static const struct dmi_system_id cht_max98090_quirk_table[] = {
+	{
+		/* Banjo model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Candy model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Candy"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{
 		/* Clapper model Chromebook */
 		.matches = {
@@ -406,6 +420,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = {
 		},
 		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
 	},
+	{
+		/* Cyan model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Enguarde model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Glimmer model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{
 		/* Gnawty model Chromebook (Acer Chromebook CB3-111) */
 		.matches = {
@@ -413,6 +448,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = {
 		},
 		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
 	},
+	{
+		/* Heli model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Heli"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Kip model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Kip"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Ninja model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Orco model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Orco"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Quawks model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Rambi model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Squawks model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Sumo model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{
 		/* Swanky model Chromebook (Toshiba Chromebook 2) */
 		.matches = {
@@ -420,6 +511,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = {
 		},
 		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
 	},
+	{
+		/* Winky model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Winky"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{}
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 22afe6242387eca1ac3905fbde7bef38deb0357d Mon Sep 17 00:00:00 2001
From: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Date: Tue, 20 Aug 2019 13:02:35 +0200
Subject: ASoC: wm8904: fix typo in DAPM kcontrol name
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Trivial fix for typo in "Capture Inverting Mux"es' name.

Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/f95ae1085f9f3c137a122c4d95728711613c15f7.1566298834.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8904.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 5ebdd1d9afde..525e4ef654a1 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -866,7 +866,7 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4,
 			    lin_text);
 
 static const struct snd_kcontrol_new lin_inv_mux =
-	SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum);
+	SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum);
 
 static const char *rin_text[] = {
 	"IN1R", "IN2R", "IN3R"
@@ -882,7 +882,7 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4,
 			    rin_text);
 
 static const struct snd_kcontrol_new rin_inv_mux =
-	SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum);
+	SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum);
 
 static const char *aif_text[] = {
 	"Left", "Right"
-- 
cgit v1.2.3-59-g8ed1b


From 0a05f2e865ad6e461cada8f931b34551287dbba2 Mon Sep 17 00:00:00 2001
From: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Date: Tue, 20 Aug 2019 13:02:37 +0200
Subject: ASoC: wm8904: implement input mode select as a mux
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Make '* Capture Mode' a mux. This makes DAPM know that in single-ended
mode only inverting mux paths need to be enabled.

Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/125cd3c9f298da9b08a4d6002d4c00d70a898950.1566298834.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8904.c | 52 ++++++++++++++++++++++++++++++-----------------
 1 file changed, 33 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 525e4ef654a1..bcb3c9d5abf0 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -545,18 +545,6 @@ static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
 static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0);
 static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 
-static const char *input_mode_text[] = {
-	"Single-Ended", "Differential Line", "Differential Mic"
-};
-
-static SOC_ENUM_SINGLE_DECL(lin_mode,
-			    WM8904_ANALOGUE_LEFT_INPUT_1, 0,
-			    input_mode_text);
-
-static SOC_ENUM_SINGLE_DECL(rin_mode,
-			    WM8904_ANALOGUE_RIGHT_INPUT_1, 0,
-			    input_mode_text);
-
 static const char *hpf_mode_text[] = {
 	"Hi-fi", "Voice 1", "Voice 2", "Voice 3"
 };
@@ -591,9 +579,6 @@ static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = {
 SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT,
 		 WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv),
 
-SOC_ENUM("Left Capture Mode", lin_mode),
-SOC_ENUM("Right Capture Mode", rin_mode),
-
 /* No TLV since it depends on mode */
 SOC_DOUBLE_R("Capture Volume", WM8904_ANALOGUE_LEFT_INPUT_0,
 	     WM8904_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0),
@@ -852,6 +837,10 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static const char *input_mode_text[] = {
+	"Single-Ended", "Differential Line", "Differential Mic"
+};
+
 static const char *lin_text[] = {
 	"IN1L", "IN2L", "IN3L"
 };
@@ -868,6 +857,13 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4,
 static const struct snd_kcontrol_new lin_inv_mux =
 	SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum);
 
+static SOC_ENUM_SINGLE_DECL(lin_mode_enum,
+			    WM8904_ANALOGUE_LEFT_INPUT_1, 0,
+			    input_mode_text);
+
+static const struct snd_kcontrol_new lin_mode =
+	SOC_DAPM_ENUM("Left Capture Mode", lin_mode_enum);
+
 static const char *rin_text[] = {
 	"IN1R", "IN2R", "IN3R"
 };
@@ -884,6 +880,13 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4,
 static const struct snd_kcontrol_new rin_inv_mux =
 	SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum);
 
+static SOC_ENUM_SINGLE_DECL(rin_mode_enum,
+			    WM8904_ANALOGUE_RIGHT_INPUT_1, 0,
+			    input_mode_text);
+
+static const struct snd_kcontrol_new rin_mode =
+	SOC_DAPM_ENUM("Right Capture Mode", rin_mode_enum);
+
 static const char *aif_text[] = {
 	"Left", "Right"
 };
@@ -932,9 +935,11 @@ SND_SOC_DAPM_SUPPLY("MICBIAS", WM8904_MIC_BIAS_CONTROL_0, 0, 0, NULL, 0),
 SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lin_mux),
 SND_SOC_DAPM_MUX("Left Capture Inverting Mux", SND_SOC_NOPM, 0, 0,
 		 &lin_inv_mux),
+SND_SOC_DAPM_MUX("Left Capture Mode", SND_SOC_NOPM, 0, 0, &lin_mode),
 SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rin_mux),
 SND_SOC_DAPM_MUX("Right Capture Inverting Mux", SND_SOC_NOPM, 0, 0,
 		 &rin_inv_mux),
+SND_SOC_DAPM_MUX("Right Capture Mode", SND_SOC_NOPM, 0, 0, &rin_mode),
 
 SND_SOC_DAPM_PGA("Left Capture PGA", WM8904_POWER_MANAGEMENT_0, 1, 0,
 		 NULL, 0),
@@ -1057,6 +1062,12 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
 	{ "Left Capture Inverting Mux", "IN2L", "IN2L" },
 	{ "Left Capture Inverting Mux", "IN3L", "IN3L" },
 
+	{ "Left Capture Mode", "Single-Ended", "Left Capture Inverting Mux" },
+	{ "Left Capture Mode", "Differential Line", "Left Capture Mux" },
+	{ "Left Capture Mode", "Differential Line", "Left Capture Inverting Mux" },
+	{ "Left Capture Mode", "Differential Mic", "Left Capture Mux" },
+	{ "Left Capture Mode", "Differential Mic", "Left Capture Inverting Mux" },
+
 	{ "Right Capture Mux", "IN1R", "IN1R" },
 	{ "Right Capture Mux", "IN2R", "IN2R" },
 	{ "Right Capture Mux", "IN3R", "IN3R" },
@@ -1065,11 +1076,14 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
 	{ "Right Capture Inverting Mux", "IN2R", "IN2R" },
 	{ "Right Capture Inverting Mux", "IN3R", "IN3R" },
 
-	{ "Left Capture PGA", NULL, "Left Capture Mux" },
-	{ "Left Capture PGA", NULL, "Left Capture Inverting Mux" },
+	{ "Right Capture Mode", "Single-Ended", "Right Capture Inverting Mux" },
+	{ "Right Capture Mode", "Differential Line", "Right Capture Mux" },
+	{ "Right Capture Mode", "Differential Line", "Right Capture Inverting Mux" },
+	{ "Right Capture Mode", "Differential Mic", "Right Capture Mux" },
+	{ "Right Capture Mode", "Differential Mic", "Right Capture Inverting Mux" },
 
-	{ "Right Capture PGA", NULL, "Right Capture Mux" },
-	{ "Right Capture PGA", NULL, "Right Capture Inverting Mux" },
+	{ "Left Capture PGA", NULL, "Left Capture Mode" },
+	{ "Right Capture PGA", NULL, "Right Capture Mode" },
 
 	{ "AIFOUTL Mux", "Left", "ADCL" },
 	{ "AIFOUTL Mux", "Right", "ADCR" },
-- 
cgit v1.2.3-59-g8ed1b


From bf283a05c09b58db83afbb1a8a3c6a684c56c1bb Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:08 +0200
Subject: ASoC: sun4i-i2s: Register regmap and PCM before our component

So far the regmap and the dmaengine PCM are registered after our component
has been, which means that our driver isn't properly initialised by then.

Let's fix that.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/67e303f37f141ef73ce9ed47d7f831b63c694424.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 7fa5c61169db..85c3b2c8cd77 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1148,11 +1148,9 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 			goto err_pm_disable;
 	}
 
-	ret = devm_snd_soc_register_component(&pdev->dev,
-					      &sun4i_i2s_component,
-					      &sun4i_i2s_dai, 1);
+	ret = sun4i_i2s_init_regmap_fields(&pdev->dev, i2s);
 	if (ret) {
-		dev_err(&pdev->dev, "Could not register DAI\n");
+		dev_err(&pdev->dev, "Could not initialise regmap fields\n");
 		goto err_suspend;
 	}
 
@@ -1162,9 +1160,11 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 		goto err_suspend;
 	}
 
-	ret = sun4i_i2s_init_regmap_fields(&pdev->dev, i2s);
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &sun4i_i2s_component,
+					      &sun4i_i2s_dai, 1);
 	if (ret) {
-		dev_err(&pdev->dev, "Could not initialise regmap fields\n");
+		dev_err(&pdev->dev, "Could not register DAI\n");
 		goto err_suspend;
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From a49d24e7d8d4fd4edf59e6373983e0bf4a2cca15 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:09 +0200
Subject: ASoC: sun4i-i2s: Switch to devm for PCM register

Since the introduction of the driver, a new managed helper for the
dmaengine PCM registration has been created. Let's use it to simplify a bit
our probe and remove functions.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/606d271187091e858e8c15e20555af0b79798fe1.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index d97d694c48df..70608fa30bf2 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1154,7 +1154,7 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
 		goto err_suspend;
 	}
 
-	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register PCM\n");
 		goto err_suspend;
@@ -1183,8 +1183,6 @@ static int sun4i_i2s_remove(struct platform_device *pdev)
 {
 	struct sun4i_i2s *i2s = dev_get_drvdata(&pdev->dev);
 
-	snd_dmaengine_pcm_unregister(&pdev->dev);
-
 	pm_runtime_disable(&pdev->dev);
 	if (!pm_runtime_status_suspended(&pdev->dev))
 		sun4i_i2s_runtime_suspend(&pdev->dev);
-- 
cgit v1.2.3-59-g8ed1b


From 7df8f9a20196072162d9dc8fe99943f2d35f23d5 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:14 +0200
Subject: ASoC: sun4i-i2s: Don't use the oversample to calculate BCLK

The BCLK divider should be calculated using the parameters that actually
make the BCLK rate: the number of channels, the sampling rate and the
sample width.

We've been using the oversample_rate previously because in the former SoCs,
the BCLK's parent is MCLK, which in turn is being used to generate the
oversample rate, so we end up with something like this:

oversample = mclk_rate / sampling_rate
bclk_div = oversample / word_size / channels

So, bclk_div = mclk_rate / sampling_rate / word_size / channels.

And this is actually better, since the oversampling ratio only plays a role
because the MCLK is its parent, not because of what BCLK is supposed to be.

Furthermore, that assumption of MCLK being the parent has been broken on
newer SoCs, so let's use the proper formula, and have the parent rate as an
argument.

Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3")
Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Fixes: 66ecce332538 ("ASoC: sun4i-i2s: Add compatibility with A64 codec I2S")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/c3595e3a9788c2ef2dcc30aa3c8c4953bb5cc249.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 70608fa30bf2..d879db581073 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -222,10 +222,11 @@ static const struct sun4i_i2s_clk_div sun4i_i2s_mclk_div[] = {
 };
 
 static int sun4i_i2s_get_bclk_div(struct sun4i_i2s *i2s,
-				  unsigned int oversample_rate,
+				  unsigned long parent_rate,
+				  unsigned int sampling_rate,
 				  unsigned int word_size)
 {
-	int div = oversample_rate / word_size / 2;
+	int div = parent_rate / sampling_rate / word_size / 2;
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(sun4i_i2s_bclk_div); i++) {
@@ -315,8 +316,8 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	bclk_div = sun4i_i2s_get_bclk_div(i2s, oversample_rate,
-					  word_size);
+	bclk_div = sun4i_i2s_get_bclk_div(i2s, i2s->mclk_freq,
+					  rate, word_size);
 	if (bclk_div < 0) {
 		dev_err(dai->dev, "Unsupported BCLK divider: %d\n", bclk_div);
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From c7dd0828c088a71f30de8d249f63b2fa9f0d322d Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:10 +0200
Subject: ASoC: sun4i-i2s: Replace call to params_channels by local variable

The sun4i_i2s_hw_params already has a variable holding the value returned
by params_channels, so let's just use that variable instead of calling
params_channels multiple times.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/c0faaac69ad40248f24eed3c3b2fa1ccc4a85b70.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index d879db581073..77b7b81daf74 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -412,10 +412,9 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	/* Configure the channels */
 	regmap_field_write(i2s->field_txchansel,
-			   SUN4I_I2S_CHAN_SEL(params_channels(params)));
-
+			   SUN4I_I2S_CHAN_SEL(channels));
 	regmap_field_write(i2s->field_rxchansel,
-			   SUN4I_I2S_CHAN_SEL(params_channels(params)));
+			   SUN4I_I2S_CHAN_SEL(channels));
 
 	if (i2s->variant->has_chsel_tx_chen)
 		regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
-- 
cgit v1.2.3-59-g8ed1b


From 8bcf62b73e5421df94deca95d7d7c152997fe5b4 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:13 +0200
Subject: ASoC: sun4i-i2s: Rework MCLK divider calculation

The MCLK divider calculation is currently computing the ideal divider using
the oversample rate, the sample rate and the parent rate.

However, since we have access to the frequency is supposed to be running at
already, and as it turns out we're using it to compute the oversample rate,
we can just use the ratio between the parent rate and the MCLK rate to
simplify a bit the formula.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/dcc5deb2eb650758d268bddd20f60ba58856d024.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 77b7b81daf74..0a5fb9d4b289 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -240,11 +240,10 @@ static int sun4i_i2s_get_bclk_div(struct sun4i_i2s *i2s,
 }
 
 static int sun4i_i2s_get_mclk_div(struct sun4i_i2s *i2s,
-				  unsigned int oversample_rate,
-				  unsigned int module_rate,
-				  unsigned int sampling_rate)
+				  unsigned long parent_rate,
+				  unsigned long mclk_rate)
 {
-	int div = module_rate / sampling_rate / oversample_rate;
+	int div = parent_rate / mclk_rate;
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(sun4i_i2s_mclk_div); i++) {
@@ -323,8 +322,7 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	mclk_div = sun4i_i2s_get_mclk_div(i2s, oversample_rate,
-					  clk_rate, rate);
+	mclk_div = sun4i_i2s_get_mclk_div(i2s, clk_rate, i2s->mclk_freq);
 	if (mclk_div < 0) {
 		dev_err(dai->dev, "Unsupported MCLK divider: %d\n", mclk_div);
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From d5e120422db8808e1c8b1507900ca393a877c58f Mon Sep 17 00:00:00 2001
From: Daniel Stuart <daniel.stuart14@gmail.com>
Date: Thu, 15 Aug 2019 14:12:55 -0300
Subject: ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need
 pmc_plt_clk_0 quirk

Every single baytrail chromebook sets PMC to 0, as can be seeing
below by searching through coreboot source code:
	$ grep -rl "PMC_PLT_CLK\[0\]" .
	./rambi/variants/glimmer/devicetree.cb
	./rambi/variants/clapper/devicetree.cb
	./rambi/variants/swanky/devicetree.cb
	./rambi/variants/enguarde/devicetree.cb
	./rambi/variants/winky/devicetree.cb
	./rambi/variants/kip/devicetree.cb
	./rambi/variants/squawks/devicetree.cb
	./rambi/variants/orco/devicetree.cb
	./rambi/variants/ninja/devicetree.cb
	./rambi/variants/heli/devicetree.cb
	./rambi/variants/sumo/devicetree.cb
	./rambi/variants/banjo/devicetree.cb
	./rambi/variants/candy/devicetree.cb
	./rambi/variants/gnawty/devicetree.cb
	./rambi/variants/rambi/devicetree.cb
	./rambi/variants/quawks/devicetree.cb

Plus, Cyan (only non-baytrail chromebook with max98090) also needs
this patch for audio to work.

Thus, this commit adds all the missing devices to bsw_max98090 quirk
table, implemented by commit a182ecd3809c ("ASoC: intel:
cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0").

Signed-off-by: Daniel Stuart <daniel.stuart14@gmail.com>
Link: https://lore.kernel.org/r/20190815171300.30126-1-daniel.stuart14@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++++++++++
 1 file changed, 98 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 1db9a95e6a79..eaf3e2208a06 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -398,6 +398,20 @@ static struct snd_soc_card snd_soc_card_cht = {
 };
 
 static const struct dmi_system_id cht_max98090_quirk_table[] = {
+	{
+		/* Banjo model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Candy model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Candy"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{
 		/* Clapper model Chromebook */
 		.matches = {
@@ -405,6 +419,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = {
 		},
 		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
 	},
+	{
+		/* Cyan model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Enguarde model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Glimmer model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{
 		/* Gnawty model Chromebook (Acer Chromebook CB3-111) */
 		.matches = {
@@ -412,6 +447,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = {
 		},
 		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
 	},
+	{
+		/* Heli model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Heli"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Kip model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Kip"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Ninja model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Orco model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Orco"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Quawks model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Rambi model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Squawks model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{
+		/* Sumo model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{
 		/* Swanky model Chromebook (Toshiba Chromebook 2) */
 		.matches = {
@@ -419,6 +510,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = {
 		},
 		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
 	},
+	{
+		/* Winky model Chromebook */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Winky"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
 	{}
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From d70be625f25af7a2bc91b7d17d205f6071f08f2f Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:11 +0200
Subject: ASoC: sun4i-i2s: Move the channel configuration to a callback

The two main generations of our I2S controller require a slightly different
channel configuration, mostly because of a quite different register layout
and some additional registers being needed on the newer generation.

This used to be controlled through a bunch of booleans, however this proved
to be quite impractical, especially since a bunch of SoCs forgot to set
those parameters and therefore were broken from that point of view.

Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/6414463de69584e8227fa495b13aa5f4798e1f0e.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 156 ++++++++++++++++++++------------------------
 1 file changed, 69 insertions(+), 87 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 2c909c6cafa9..42e45c9a947a 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -80,6 +80,7 @@
 #define SUN4I_I2S_TX_CNT_REG		0x2c
 
 #define SUN4I_I2S_TX_CHAN_SEL_REG	0x30
+#define SUN4I_I2S_CHAN_SEL_MASK			GENMASK(2, 0)
 #define SUN4I_I2S_CHAN_SEL(num_chan)		(((num_chan) - 1) << 0)
 
 #define SUN4I_I2S_TX_CHAN_MAP_REG	0x34
@@ -122,8 +123,6 @@ struct sun4i_i2s;
  * @has_reset: SoC needs reset deasserted.
  * @has_slave_select_bit: SoC has a bit to enable slave mode.
  * @has_fmt_set_lrck_period: SoC requires lrclk period to be set.
- * @has_chcfg: tx and rx slot number need to be set.
- * @has_chsel_tx_chen: SoC requires that the tx channels are enabled.
  * @has_chsel_offset: SoC uses offset for selecting dai operational mode.
  * @reg_offset_txdata: offset of the tx fifo.
  * @sun4i_i2s_regmap: regmap config to use.
@@ -135,17 +134,11 @@ struct sun4i_i2s;
  * @field_fmt_bclk: regmap field to set clk polarity.
  * @field_fmt_lrclk: regmap field to set frame polarity.
  * @field_fmt_mode: regmap field to set the operational mode.
- * @field_txchanmap: location of the tx channel mapping register.
- * @field_rxchanmap: location of the rx channel mapping register.
- * @field_txchansel: location of the tx channel select bit fields.
- * @field_rxchansel: location of the rx channel select bit fields.
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
 	bool				has_slave_select_bit;
 	bool				has_fmt_set_lrck_period;
-	bool				has_chcfg;
-	bool				has_chsel_tx_chen;
 	bool				has_chsel_offset;
 	unsigned int			reg_offset_txdata;	/* TX FIFO */
 	const struct regmap_config	*sun4i_i2s_regmap;
@@ -159,13 +152,11 @@ struct sun4i_i2s_quirks {
 	struct reg_field		field_fmt_bclk;
 	struct reg_field		field_fmt_lrclk;
 	struct reg_field		field_fmt_mode;
-	struct reg_field		field_txchanmap;
-	struct reg_field		field_rxchanmap;
-	struct reg_field		field_txchansel;
-	struct reg_field		field_rxchansel;
 
 	s8	(*get_sr)(const struct sun4i_i2s *, int);
 	s8	(*get_wss)(const struct sun4i_i2s *, int);
+	int	(*set_chan_cfg)(const struct sun4i_i2s *,
+				const struct snd_pcm_hw_params *);
 };
 
 struct sun4i_i2s {
@@ -186,10 +177,6 @@ struct sun4i_i2s {
 	struct regmap_field	*field_fmt_bclk;
 	struct regmap_field	*field_fmt_lrclk;
 	struct regmap_field	*field_fmt_mode;
-	struct regmap_field	*field_txchanmap;
-	struct regmap_field	*field_rxchanmap;
-	struct regmap_field	*field_txchansel;
-	struct regmap_field	*field_rxchansel;
 
 	const struct sun4i_i2s_quirks	*variant;
 };
@@ -380,44 +367,77 @@ static s8 sun8i_i2s_get_sr_wss(const struct sun4i_i2s *i2s, int width)
 	return (width - 8) / 4 + 1;
 }
 
-static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
-			       struct snd_pcm_hw_params *params,
-			       struct snd_soc_dai *dai)
+static int sun4i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
+				  const struct snd_pcm_hw_params *params)
 {
-	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	int sr, wss, channels;
-	u32 width;
+	unsigned int channels = params_channels(params);
 
-	channels = params_channels(params);
-	if (channels != 2) {
-		dev_err(dai->dev, "Unsupported number of channels: %d\n",
-			channels);
+	if (channels != 2)
 		return -EINVAL;
-	}
 
-	if (i2s->variant->has_chcfg) {
-		regmap_update_bits(i2s->regmap, SUN8I_I2S_CHAN_CFG_REG,
-				   SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK,
-				   SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(channels));
-		regmap_update_bits(i2s->regmap, SUN8I_I2S_CHAN_CFG_REG,
-				   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK,
-				   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(channels));
-	}
+	/* Map the channels for playback and capture */
+	regmap_write(i2s->regmap, SUN4I_I2S_TX_CHAN_MAP_REG, 0x76543210);
+	regmap_write(i2s->regmap, SUN4I_I2S_RX_CHAN_MAP_REG, 0x00003210);
+
+	/* Configure the channels */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_TX_CHAN_SEL_REG,
+			   SUN4I_I2S_CHAN_SEL_MASK,
+			   SUN4I_I2S_CHAN_SEL(channels));
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_RX_CHAN_SEL_REG,
+			   SUN4I_I2S_CHAN_SEL_MASK,
+			   SUN4I_I2S_CHAN_SEL(channels));
+
+	return 0;
+}
+
+static int sun8i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
+				  const struct snd_pcm_hw_params *params)
+{
+	unsigned int channels = params_channels(params);
+
+	if (channels != 2)
+		return -EINVAL;
 
 	/* Map the channels for playback and capture */
-	regmap_field_write(i2s->field_txchanmap, 0x76543210);
-	regmap_field_write(i2s->field_rxchanmap, 0x00003210);
+	regmap_write(i2s->regmap, SUN8I_I2S_TX_CHAN_MAP_REG, 0x76543210);
+	regmap_write(i2s->regmap, SUN8I_I2S_RX_CHAN_MAP_REG, 0x76543210);
 
 	/* Configure the channels */
-	regmap_field_write(i2s->field_txchansel,
+	regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
+			   SUN4I_I2S_CHAN_SEL_MASK,
 			   SUN4I_I2S_CHAN_SEL(channels));
-	regmap_field_write(i2s->field_rxchansel,
+
+	regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG,
+			   SUN4I_I2S_CHAN_SEL_MASK,
 			   SUN4I_I2S_CHAN_SEL(channels));
 
-	if (i2s->variant->has_chsel_tx_chen)
-		regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
-				   SUN8I_I2S_TX_CHAN_EN_MASK,
-				   SUN8I_I2S_TX_CHAN_EN(channels));
+	regmap_update_bits(i2s->regmap, SUN8I_I2S_CHAN_CFG_REG,
+			   SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK,
+			   SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(channels));
+	regmap_update_bits(i2s->regmap, SUN8I_I2S_CHAN_CFG_REG,
+			   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK,
+			   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(channels));
+
+	regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
+			   SUN8I_I2S_TX_CHAN_EN_MASK,
+			   SUN8I_I2S_TX_CHAN_EN(channels));
+
+	return 0;
+}
+
+static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
+			       struct snd_pcm_hw_params *params,
+			       struct snd_soc_dai *dai)
+{
+	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	int ret, sr, wss;
+	u32 width;
+
+	ret = i2s->variant->set_chan_cfg(i2s, params);
+	if (ret < 0) {
+		dev_err(dai->dev, "Invalid channel configuration\n");
+		return ret;
+	}
 
 	switch (params_physical_width(params)) {
 	case 16:
@@ -915,12 +935,9 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.has_slave_select_bit	= true,
 	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
-	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
-	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
-	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
-	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
+	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
 };
 
 static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
@@ -934,12 +951,9 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.has_slave_select_bit	= true,
 	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
-	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
-	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
-	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
-	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
+	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
 };
 
 static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
@@ -953,12 +967,9 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.has_slave_select_bit	= true,
 	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
-	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
-	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
-	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
-	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
+	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
 };
 
 static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
@@ -968,8 +979,6 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.mclk_offset		= 1,
 	.bclk_offset		= 2,
 	.has_fmt_set_lrck_period = true,
-	.has_chcfg		= true,
-	.has_chsel_tx_chen	= true,
 	.has_chsel_offset	= true,
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
@@ -977,12 +986,9 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 19, 19),
 	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_CTRL_REG, 4, 5),
-	.field_txchanmap	= REG_FIELD(SUN8I_I2S_TX_CHAN_MAP_REG, 0, 31),
-	.field_rxchanmap	= REG_FIELD(SUN8I_I2S_RX_CHAN_MAP_REG, 0, 31),
-	.field_txchansel	= REG_FIELD(SUN8I_I2S_TX_CHAN_SEL_REG, 0, 2),
-	.field_rxchansel	= REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2),
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
+	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
 };
 
 static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
@@ -996,12 +1002,9 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
-	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
-	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
-	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
-	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
+	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
 };
 
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
@@ -1043,28 +1046,7 @@ static int sun4i_i2s_init_regmap_fields(struct device *dev,
 	if (IS_ERR(i2s->field_fmt_mode))
 		return PTR_ERR(i2s->field_fmt_mode);
 
-	i2s->field_txchanmap =
-			devm_regmap_field_alloc(dev, i2s->regmap,
-						i2s->variant->field_txchanmap);
-	if (IS_ERR(i2s->field_txchanmap))
-		return PTR_ERR(i2s->field_txchanmap);
-
-	i2s->field_rxchanmap =
-			devm_regmap_field_alloc(dev, i2s->regmap,
-						i2s->variant->field_rxchanmap);
-	if (IS_ERR(i2s->field_rxchanmap))
-		return PTR_ERR(i2s->field_rxchanmap);
-
-	i2s->field_txchansel =
-			devm_regmap_field_alloc(dev, i2s->regmap,
-						i2s->variant->field_txchansel);
-	if (IS_ERR(i2s->field_txchansel))
-		return PTR_ERR(i2s->field_txchansel);
-
-	i2s->field_rxchansel =
-			devm_regmap_field_alloc(dev, i2s->regmap,
-						i2s->variant->field_rxchansel);
-	return PTR_ERR_OR_ZERO(i2s->field_rxchansel);
+	return 0;
 }
 
 static int sun4i_i2s_probe(struct platform_device *pdev)
-- 
cgit v1.2.3-59-g8ed1b


From 71137bcd0a9a778f9407a3bee46c62fcccee4f83 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:12 +0200
Subject: ASoC: sun4i-i2s: Move the format configuration to a callback

The two main generations of our I2S controller require a slightly different
format configuration, mostly because of a quite different register layout
and some additional registers being needed on the newer generation.

This used to be controlled through a bunch of booleans, however this proved
to be quite impractical, especially since a bunch of SoCs forgot to set
those parameters and therefore were broken from that point of view.

Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/dc818644c3e40734e7a97247c994b1fca1c3c047.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 199 +++++++++++++++++++++++---------------------
 1 file changed, 106 insertions(+), 93 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 42e45c9a947a..93ea627e2f1f 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -93,6 +93,11 @@
 #define SUN8I_I2S_CTRL_BCLK_OUT			BIT(18)
 #define SUN8I_I2S_CTRL_LRCK_OUT			BIT(17)
 
+#define SUN8I_I2S_CTRL_MODE_MASK		GENMASK(5, 4)
+#define SUN8I_I2S_CTRL_MODE_RIGHT		(2 << 4)
+#define SUN8I_I2S_CTRL_MODE_LEFT		(1 << 4)
+#define SUN8I_I2S_CTRL_MODE_PCM			(0 << 4)
+
 #define SUN8I_I2S_FMT0_LRCK_PERIOD_MASK		GENMASK(17, 8)
 #define SUN8I_I2S_FMT0_LRCK_PERIOD(period)	((period - 1) << 8)
 
@@ -121,9 +126,7 @@ struct sun4i_i2s;
  * struct sun4i_i2s_quirks - Differences between SoC variants.
  *
  * @has_reset: SoC needs reset deasserted.
- * @has_slave_select_bit: SoC has a bit to enable slave mode.
  * @has_fmt_set_lrck_period: SoC requires lrclk period to be set.
- * @has_chsel_offset: SoC uses offset for selecting dai operational mode.
  * @reg_offset_txdata: offset of the tx fifo.
  * @sun4i_i2s_regmap: regmap config to use.
  * @mclk_offset: Value by which mclkdiv needs to be adjusted.
@@ -133,13 +136,10 @@ struct sun4i_i2s;
  * @field_fmt_sr: regmap field to set sample resolution.
  * @field_fmt_bclk: regmap field to set clk polarity.
  * @field_fmt_lrclk: regmap field to set frame polarity.
- * @field_fmt_mode: regmap field to set the operational mode.
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
-	bool				has_slave_select_bit;
 	bool				has_fmt_set_lrck_period;
-	bool				has_chsel_offset;
 	unsigned int			reg_offset_txdata;	/* TX FIFO */
 	const struct regmap_config	*sun4i_i2s_regmap;
 	unsigned int			mclk_offset;
@@ -151,12 +151,12 @@ struct sun4i_i2s_quirks {
 	struct reg_field		field_fmt_sr;
 	struct reg_field		field_fmt_bclk;
 	struct reg_field		field_fmt_lrclk;
-	struct reg_field		field_fmt_mode;
 
 	s8	(*get_sr)(const struct sun4i_i2s *, int);
 	s8	(*get_wss)(const struct sun4i_i2s *, int);
 	int	(*set_chan_cfg)(const struct sun4i_i2s *,
 				const struct snd_pcm_hw_params *);
+	int	(*set_fmt)(const struct sun4i_i2s *, unsigned int);
 };
 
 struct sun4i_i2s {
@@ -176,7 +176,6 @@ struct sun4i_i2s {
 	struct regmap_field	*field_fmt_sr;
 	struct regmap_field	*field_fmt_bclk;
 	struct regmap_field	*field_fmt_lrclk;
-	struct regmap_field	*field_fmt_mode;
 
 	const struct sun4i_i2s_quirks	*variant;
 };
@@ -465,52 +464,117 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 				      params_width(params));
 }
 
-static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+static int sun4i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
+				 unsigned int fmt)
 {
-	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	u32 val;
-	u32 offset = 0;
-	u32 bclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
-	u32 lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
 
 	/* DAI Mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		val = SUN4I_I2S_FMT0_FMT_I2S;
-		offset = 1;
 		break;
+
 	case SND_SOC_DAIFMT_LEFT_J:
 		val = SUN4I_I2S_FMT0_FMT_LEFT_J;
 		break;
+
 	case SND_SOC_DAIFMT_RIGHT_J:
 		val = SUN4I_I2S_FMT0_FMT_RIGHT_J;
 		break;
+
 	default:
-		dev_err(dai->dev, "Unsupported format: %d\n",
-			fmt & SND_SOC_DAIFMT_FORMAT_MASK);
 		return -EINVAL;
 	}
 
-	if (i2s->variant->has_chsel_offset) {
-		/*
-		 * offset being set indicates that we're connected to an i2s
-		 * device, however offset is only used on the sun8i block and
-		 * i2s shares the same setting with the LJ format. Increment
-		 * val so that the bit to value to write is correct.
-		 */
-		if (offset > 0)
-			val++;
-		/* blck offset determines whether i2s or LJ */
-		regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
-				   SUN8I_I2S_TX_CHAN_OFFSET_MASK,
-				   SUN8I_I2S_TX_CHAN_OFFSET(offset));
-
-		regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG,
-				   SUN8I_I2S_TX_CHAN_OFFSET_MASK,
-				   SUN8I_I2S_TX_CHAN_OFFSET(offset));
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
+			   SUN4I_I2S_FMT0_FMT_MASK, val);
+
+	/* DAI clock master masks */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/* BCLK and LRCLK master */
+		val = SUN4I_I2S_CTRL_MODE_MASTER;
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+		/* BCLK and LRCLK slave */
+		val = SUN4I_I2S_CTRL_MODE_SLAVE;
+		break;
+
+	default:
+		return -EINVAL;
 	}
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_MODE_MASK, val);
+	return 0;
+}
 
-	regmap_field_write(i2s->field_fmt_mode, val);
+static int sun8i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
+				 unsigned int fmt)
+{
+	u32 mode, val;
+	u8 offset;
+
+	/* DAI Mode */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		mode = SUN8I_I2S_CTRL_MODE_LEFT;
+		offset = 1;
+		break;
+
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode = SUN8I_I2S_CTRL_MODE_LEFT;
+		offset = 0;
+		break;
+
+	case SND_SOC_DAIFMT_RIGHT_J:
+		mode = SUN8I_I2S_CTRL_MODE_RIGHT;
+		offset = 0;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN8I_I2S_CTRL_MODE_MASK, mode);
+	regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
+			   SUN8I_I2S_TX_CHAN_OFFSET_MASK,
+			   SUN8I_I2S_TX_CHAN_OFFSET(offset));
+	regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG,
+			   SUN8I_I2S_TX_CHAN_OFFSET_MASK,
+			   SUN8I_I2S_TX_CHAN_OFFSET(offset));
+
+	/* DAI clock master masks */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/* BCLK and LRCLK master */
+		val = SUN8I_I2S_CTRL_BCLK_OUT |	SUN8I_I2S_CTRL_LRCK_OUT;
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+		/* BCLK and LRCLK slave */
+		val = 0;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN8I_I2S_CTRL_BCLK_OUT | SUN8I_I2S_CTRL_LRCK_OUT,
+			   val);
+
+	return 0;
+}
+
+static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	u32 bclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
+	u32 lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
+	int ret;
 
 	/* DAI clock polarity */
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -538,50 +602,10 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	regmap_field_write(i2s->field_fmt_bclk, bclk_polarity);
 	regmap_field_write(i2s->field_fmt_lrclk, lrclk_polarity);
 
-	if (i2s->variant->has_slave_select_bit) {
-		/* DAI clock master masks */
-		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-		case SND_SOC_DAIFMT_CBS_CFS:
-			/* BCLK and LRCLK master */
-			val = SUN4I_I2S_CTRL_MODE_MASTER;
-			break;
-		case SND_SOC_DAIFMT_CBM_CFM:
-			/* BCLK and LRCLK slave */
-			val = SUN4I_I2S_CTRL_MODE_SLAVE;
-			break;
-		default:
-			dev_err(dai->dev, "Unsupported slave setting: %d\n",
-				fmt & SND_SOC_DAIFMT_MASTER_MASK);
-			return -EINVAL;
-		}
-		regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-				   SUN4I_I2S_CTRL_MODE_MASK,
-				   val);
-	} else {
-		/*
-		 * The newer i2s block does not have a slave select bit,
-		 * instead the clk pins are configured as inputs.
-		 */
-		/* DAI clock master masks */
-		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-		case SND_SOC_DAIFMT_CBS_CFS:
-			/* BCLK and LRCLK master */
-			val = SUN8I_I2S_CTRL_BCLK_OUT |
-				SUN8I_I2S_CTRL_LRCK_OUT;
-			break;
-		case SND_SOC_DAIFMT_CBM_CFM:
-			/* BCLK and LRCLK slave */
-			val = 0;
-			break;
-		default:
-			dev_err(dai->dev, "Unsupported slave setting: %d\n",
-				fmt & SND_SOC_DAIFMT_MASTER_MASK);
-			return -EINVAL;
-		}
-		regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-				   SUN8I_I2S_CTRL_BCLK_OUT |
-				   SUN8I_I2S_CTRL_LRCK_OUT,
-				   val);
+	ret = i2s->variant->set_fmt(i2s, fmt);
+	if (ret) {
+		dev_err(dai->dev, "Unsupported format configuration\n");
+		return ret;
 	}
 
 	/* Set significant bits in our FIFOs */
@@ -933,11 +957,10 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
-	.has_slave_select_bit	= true,
-	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
 	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
+	.set_fmt		= sun4i_i2s_set_soc_fmt,
 };
 
 static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
@@ -949,11 +972,10 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
-	.has_slave_select_bit	= true,
-	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
 	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
+	.set_fmt		= sun4i_i2s_set_soc_fmt,
 };
 
 static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
@@ -965,11 +987,10 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
-	.has_slave_select_bit	= true,
-	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
 	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
+	.set_fmt		= sun8i_i2s_set_soc_fmt,
 };
 
 static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
@@ -979,32 +1000,30 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.mclk_offset		= 1,
 	.bclk_offset		= 2,
 	.has_fmt_set_lrck_period = true,
-	.has_chsel_offset	= true,
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 19, 19),
-	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_CTRL_REG, 4, 5),
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
 	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
+	.set_fmt		= sun8i_i2s_set_soc_fmt,
 };
 
 static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
-	.has_slave_select_bit	= true,
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
-	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
 	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
+	.set_fmt		= sun4i_i2s_set_soc_fmt,
 };
 
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
@@ -1040,12 +1059,6 @@ static int sun4i_i2s_init_regmap_fields(struct device *dev,
 	if (IS_ERR(i2s->field_fmt_lrclk))
 		return PTR_ERR(i2s->field_fmt_lrclk);
 
-	i2s->field_fmt_mode =
-			devm_regmap_field_alloc(dev, i2s->regmap,
-						i2s->variant->field_fmt_mode);
-	if (IS_ERR(i2s->field_fmt_mode))
-		return PTR_ERR(i2s->field_fmt_mode);
-
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From fb19739d7f688142b61d0fca476188c4fd9e937a Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:15 +0200
Subject: ASoC: sun4i-i2s: Use module clock as BCLK parent on newer SoCs

On the first generation of Allwinner SoCs (A10-A31), the i2s controller was
using the MCLK as BCLK parent. However, this changed since the introduction
of the A83t and BCLK now uses the module clock as its parent.

Let's introduce a hook to get the parent rate and use that in our divider
calculations.

Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3")
Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/0b6665be216b3bd0e7bc43724818f05f3f8ee881.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 21 +++++++++++++++++++--
 1 file changed, 19 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 93ea627e2f1f..acfcdb26086a 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -152,6 +152,7 @@ struct sun4i_i2s_quirks {
 	struct reg_field		field_fmt_bclk;
 	struct reg_field		field_fmt_lrclk;
 
+	unsigned long (*get_bclk_parent_rate)(const struct sun4i_i2s *);
 	s8	(*get_sr)(const struct sun4i_i2s *, int);
 	s8	(*get_wss)(const struct sun4i_i2s *, int);
 	int	(*set_chan_cfg)(const struct sun4i_i2s *,
@@ -207,6 +208,16 @@ static const struct sun4i_i2s_clk_div sun4i_i2s_mclk_div[] = {
 	/* TODO - extend divide ratio supported by newer SoCs */
 };
 
+static unsigned long sun4i_i2s_get_bclk_parent_rate(const struct sun4i_i2s *i2s)
+{
+	return i2s->mclk_freq;
+}
+
+static unsigned long sun8i_i2s_get_bclk_parent_rate(const struct sun4i_i2s *i2s)
+{
+	return clk_get_rate(i2s->mod_clk);
+}
+
 static int sun4i_i2s_get_bclk_div(struct sun4i_i2s *i2s,
 				  unsigned long parent_rate,
 				  unsigned int sampling_rate,
@@ -259,7 +270,7 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 				  unsigned int word_size)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	unsigned int oversample_rate, clk_rate;
+	unsigned int oversample_rate, clk_rate, bclk_parent_rate;
 	int bclk_div, mclk_div;
 	int ret;
 
@@ -301,7 +312,8 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	bclk_div = sun4i_i2s_get_bclk_div(i2s, i2s->mclk_freq,
+	bclk_parent_rate = i2s->variant->get_bclk_parent_rate(i2s);
+	bclk_div = sun4i_i2s_get_bclk_div(i2s, bclk_parent_rate,
 					  rate, word_size);
 	if (bclk_div < 0) {
 		dev_err(dai->dev, "Unsupported BCLK divider: %d\n", bclk_div);
@@ -957,6 +969,7 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.get_bclk_parent_rate	= sun4i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
 	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
@@ -972,6 +985,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.get_bclk_parent_rate	= sun4i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
 	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
@@ -987,6 +1001,7 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.get_bclk_parent_rate	= sun8i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
 	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
@@ -1005,6 +1020,7 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 19, 19),
+	.get_bclk_parent_rate	= sun8i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
 	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
@@ -1020,6 +1036,7 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.get_bclk_parent_rate	= sun4i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
 	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
-- 
cgit v1.2.3-59-g8ed1b


From c1d3a921d72bd21f266ca28c15213fbe78160a4b Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:16 +0200
Subject: ASoC: sun4i-i2s: Fix the MCLK and BCLK dividers on newer SoCs

From: Marcus Cooper <codekipper@gmail.com>

The clock division dividers have changed between the older (A10/A31) and
newer (H3, A64, etc) SoCs.

While this was addressed through an offset on some SoCs, it was missing
some dividers as well, so the support wasn't perfect. Let's introduce a
pointer in the quirk structure for the divider calculation functions to use
so we can have the proper range now.

Signed-off-by: Marcus Cooper <codekipper@gmail.com>
[Maxime: Fix the commit log, use a field in the quirk structure]
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/0e5b4abf06cd3202354315201c6af44caeb20236.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 63 +++++++++++++++++++++++++++++++++++----------
 1 file changed, 49 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index acfcdb26086a..0a7f1d0f7371 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -129,8 +129,6 @@ struct sun4i_i2s;
  * @has_fmt_set_lrck_period: SoC requires lrclk period to be set.
  * @reg_offset_txdata: offset of the tx fifo.
  * @sun4i_i2s_regmap: regmap config to use.
- * @mclk_offset: Value by which mclkdiv needs to be adjusted.
- * @bclk_offset: Value by which bclkdiv needs to be adjusted.
  * @field_clkdiv_mclk_en: regmap field to enable mclk output.
  * @field_fmt_wss: regmap field to set word select size.
  * @field_fmt_sr: regmap field to set sample resolution.
@@ -142,8 +140,6 @@ struct sun4i_i2s_quirks {
 	bool				has_fmt_set_lrck_period;
 	unsigned int			reg_offset_txdata;	/* TX FIFO */
 	const struct regmap_config	*sun4i_i2s_regmap;
-	unsigned int			mclk_offset;
-	unsigned int			bclk_offset;
 
 	/* Register fields for i2s */
 	struct reg_field		field_clkdiv_mclk_en;
@@ -152,6 +148,11 @@ struct sun4i_i2s_quirks {
 	struct reg_field		field_fmt_bclk;
 	struct reg_field		field_fmt_lrclk;
 
+	const struct sun4i_i2s_clk_div	*bclk_dividers;
+	unsigned int			num_bclk_dividers;
+	const struct sun4i_i2s_clk_div	*mclk_dividers;
+	unsigned int			num_mclk_dividers;
+
 	unsigned long (*get_bclk_parent_rate)(const struct sun4i_i2s *);
 	s8	(*get_sr)(const struct sun4i_i2s *, int);
 	s8	(*get_wss)(const struct sun4i_i2s *, int);
@@ -208,6 +209,24 @@ static const struct sun4i_i2s_clk_div sun4i_i2s_mclk_div[] = {
 	/* TODO - extend divide ratio supported by newer SoCs */
 };
 
+static const struct sun4i_i2s_clk_div sun8i_i2s_clk_div[] = {
+	{ .div = 1, .val = 1 },
+	{ .div = 2, .val = 2 },
+	{ .div = 4, .val = 3 },
+	{ .div = 6, .val = 4 },
+	{ .div = 8, .val = 5 },
+	{ .div = 12, .val = 6 },
+	{ .div = 16, .val = 7 },
+	{ .div = 24, .val = 8 },
+	{ .div = 32, .val = 9 },
+	{ .div = 48, .val = 10 },
+	{ .div = 64, .val = 11 },
+	{ .div = 96, .val = 12 },
+	{ .div = 128, .val = 13 },
+	{ .div = 176, .val = 14 },
+	{ .div = 192, .val = 15 },
+};
+
 static unsigned long sun4i_i2s_get_bclk_parent_rate(const struct sun4i_i2s *i2s)
 {
 	return i2s->mclk_freq;
@@ -223,11 +242,12 @@ static int sun4i_i2s_get_bclk_div(struct sun4i_i2s *i2s,
 				  unsigned int sampling_rate,
 				  unsigned int word_size)
 {
+	const struct sun4i_i2s_clk_div *dividers = i2s->variant->bclk_dividers;
 	int div = parent_rate / sampling_rate / word_size / 2;
 	int i;
 
-	for (i = 0; i < ARRAY_SIZE(sun4i_i2s_bclk_div); i++) {
-		const struct sun4i_i2s_clk_div *bdiv = &sun4i_i2s_bclk_div[i];
+	for (i = 0; i < i2s->variant->num_bclk_dividers; i++) {
+		const struct sun4i_i2s_clk_div *bdiv = &dividers[i];
 
 		if (bdiv->div == div)
 			return bdiv->val;
@@ -240,11 +260,12 @@ static int sun4i_i2s_get_mclk_div(struct sun4i_i2s *i2s,
 				  unsigned long parent_rate,
 				  unsigned long mclk_rate)
 {
+	const struct sun4i_i2s_clk_div *dividers = i2s->variant->mclk_dividers;
 	int div = parent_rate / mclk_rate;
 	int i;
 
-	for (i = 0; i < ARRAY_SIZE(sun4i_i2s_mclk_div); i++) {
-		const struct sun4i_i2s_clk_div *mdiv = &sun4i_i2s_mclk_div[i];
+	for (i = 0; i < i2s->variant->num_mclk_dividers; i++) {
+		const struct sun4i_i2s_clk_div *mdiv = &dividers[i];
 
 		if (mdiv->div == div)
 			return mdiv->val;
@@ -326,10 +347,6 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	/* Adjust the clock division values if needed */
-	bclk_div += i2s->variant->bclk_offset;
-	mclk_div += i2s->variant->mclk_offset;
-
 	regmap_write(i2s->regmap, SUN4I_I2S_CLK_DIV_REG,
 		     SUN4I_I2S_CLK_DIV_BCLK(bclk_div) |
 		     SUN4I_I2S_CLK_DIV_MCLK(mclk_div));
@@ -969,6 +986,10 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.bclk_dividers		= sun4i_i2s_bclk_div,
+	.num_bclk_dividers	= ARRAY_SIZE(sun4i_i2s_bclk_div),
+	.mclk_dividers		= sun4i_i2s_mclk_div,
+	.num_mclk_dividers	= ARRAY_SIZE(sun4i_i2s_mclk_div),
 	.get_bclk_parent_rate	= sun4i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
@@ -985,6 +1006,10 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.bclk_dividers		= sun4i_i2s_bclk_div,
+	.num_bclk_dividers	= ARRAY_SIZE(sun4i_i2s_bclk_div),
+	.mclk_dividers		= sun4i_i2s_mclk_div,
+	.num_mclk_dividers	= ARRAY_SIZE(sun4i_i2s_mclk_div),
 	.get_bclk_parent_rate	= sun4i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
@@ -1001,6 +1026,10 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.bclk_dividers		= sun8i_i2s_clk_div,
+	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
+	.mclk_dividers		= sun8i_i2s_clk_div,
+	.num_mclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
 	.get_bclk_parent_rate	= sun8i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
@@ -1012,14 +1041,16 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun8i_i2s_regmap_config,
-	.mclk_offset		= 1,
-	.bclk_offset		= 2,
 	.has_fmt_set_lrck_period = true,
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 19, 19),
+	.bclk_dividers		= sun8i_i2s_clk_div,
+	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
+	.mclk_dividers		= sun8i_i2s_clk_div,
+	.num_mclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
 	.get_bclk_parent_rate	= sun8i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun8i_i2s_get_sr_wss,
 	.get_wss		= sun8i_i2s_get_sr_wss,
@@ -1036,6 +1067,10 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
 	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
 	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.bclk_dividers		= sun4i_i2s_bclk_div,
+	.num_bclk_dividers	= ARRAY_SIZE(sun4i_i2s_bclk_div),
+	.mclk_dividers		= sun4i_i2s_mclk_div,
+	.num_mclk_dividers	= ARRAY_SIZE(sun4i_i2s_mclk_div),
 	.get_bclk_parent_rate	= sun4i_i2s_get_bclk_parent_rate,
 	.get_sr			= sun4i_i2s_get_sr,
 	.get_wss		= sun4i_i2s_get_wss,
-- 
cgit v1.2.3-59-g8ed1b


From cf2c0e1ce9544df42170fb921f12da82dc0cc8d6 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:17 +0200
Subject: ASoC: sun4i-i2s: RX and TX counter registers are swapped

The RX and TX counters registers offset have been swapped, fix that.

Fixes: fa7c0d13cb26 ("ASoC: sunxi: Add Allwinner A10 Digital Audio driver")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/8b26477560ad5fd8f69e037b167c5e61de5c26a3.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 0a7f1d0f7371..53c95e5289f5 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -76,8 +76,8 @@
 #define SUN4I_I2S_CLK_DIV_MCLK_MASK		GENMASK(3, 0)
 #define SUN4I_I2S_CLK_DIV_MCLK(mclk)			((mclk) << 0)
 
-#define SUN4I_I2S_RX_CNT_REG		0x28
-#define SUN4I_I2S_TX_CNT_REG		0x2c
+#define SUN4I_I2S_TX_CNT_REG		0x28
+#define SUN4I_I2S_RX_CNT_REG		0x2c
 
 #define SUN4I_I2S_TX_CHAN_SEL_REG	0x30
 #define SUN4I_I2S_CHAN_SEL_MASK			GENMASK(2, 0)
-- 
cgit v1.2.3-59-g8ed1b


From 515fcfbc773632e160f4b94e8df8d278a8d704f7 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:19 +0200
Subject: ASoC: sun4i-i2s: Fix LRCK and BCLK polarity offsets on newer SoCs

The LRCK and BCLK polarity offsets on newer SoCs has been
changed, yet the driver didn't take it into account for all of them.

This was taken into account for the H3, but not the A83t. This was handled
using a reg_field for the H3.

However, the value in that field will not be the same, so reg_field is not
adapted in that case. Let's change for proper calls with the regular
values.

Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/9cbdde80a299288878e58225df4d7884e0301348.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 118 ++++++++++++++++++++++----------------------
 1 file changed, 60 insertions(+), 58 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 53c95e5289f5..e3eadfe38aaf 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -46,8 +46,6 @@
 #define SUN4I_I2S_FMT0_FMT_RIGHT_J			(2 << 0)
 #define SUN4I_I2S_FMT0_FMT_LEFT_J			(1 << 0)
 #define SUN4I_I2S_FMT0_FMT_I2S				(0 << 0)
-#define SUN4I_I2S_FMT0_POLARITY_INVERTED		(1)
-#define SUN4I_I2S_FMT0_POLARITY_NORMAL			(0)
 
 #define SUN4I_I2S_FMT1_REG		0x08
 #define SUN4I_I2S_FIFO_TX_REG		0x0c
@@ -98,8 +96,14 @@
 #define SUN8I_I2S_CTRL_MODE_LEFT		(1 << 4)
 #define SUN8I_I2S_CTRL_MODE_PCM			(0 << 4)
 
+#define SUN8I_I2S_FMT0_LRCLK_POLARITY_MASK	BIT(19)
+#define SUN8I_I2S_FMT0_LRCLK_POLARITY_INVERTED		(1 << 19)
+#define SUN8I_I2S_FMT0_LRCLK_POLARITY_NORMAL		(0 << 19)
 #define SUN8I_I2S_FMT0_LRCK_PERIOD_MASK		GENMASK(17, 8)
 #define SUN8I_I2S_FMT0_LRCK_PERIOD(period)	((period - 1) << 8)
+#define SUN8I_I2S_FMT0_BCLK_POLARITY_MASK	BIT(7)
+#define SUN8I_I2S_FMT0_BCLK_POLARITY_INVERTED		(1 << 7)
+#define SUN8I_I2S_FMT0_BCLK_POLARITY_NORMAL		(0 << 7)
 
 #define SUN8I_I2S_INT_STA_REG		0x0c
 #define SUN8I_I2S_FIFO_TX_REG		0x20
@@ -132,8 +136,6 @@ struct sun4i_i2s;
  * @field_clkdiv_mclk_en: regmap field to enable mclk output.
  * @field_fmt_wss: regmap field to set word select size.
  * @field_fmt_sr: regmap field to set sample resolution.
- * @field_fmt_bclk: regmap field to set clk polarity.
- * @field_fmt_lrclk: regmap field to set frame polarity.
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
@@ -145,8 +147,6 @@ struct sun4i_i2s_quirks {
 	struct reg_field		field_clkdiv_mclk_en;
 	struct reg_field		field_fmt_wss;
 	struct reg_field		field_fmt_sr;
-	struct reg_field		field_fmt_bclk;
-	struct reg_field		field_fmt_lrclk;
 
 	const struct sun4i_i2s_clk_div	*bclk_dividers;
 	unsigned int			num_bclk_dividers;
@@ -176,8 +176,6 @@ struct sun4i_i2s {
 	struct regmap_field	*field_clkdiv_mclk_en;
 	struct regmap_field	*field_fmt_wss;
 	struct regmap_field	*field_fmt_sr;
-	struct regmap_field	*field_fmt_bclk;
-	struct regmap_field	*field_fmt_lrclk;
 
 	const struct sun4i_i2s_quirks	*variant;
 };
@@ -498,6 +496,33 @@ static int sun4i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
 {
 	u32 val;
 
+	/* DAI clock polarity */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_IF:
+		/* Invert both clocks */
+		val = SUN4I_I2S_FMT0_BCLK_POLARITY_INVERTED |
+		      SUN4I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/* Invert bit clock */
+		val = SUN4I_I2S_FMT0_BCLK_POLARITY_INVERTED;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		/* Invert frame clock */
+		val = SUN4I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		val = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
+			   SUN4I_I2S_FMT0_LRCLK_POLARITY_MASK |
+			   SUN4I_I2S_FMT0_BCLK_POLARITY_MASK,
+			   val);
+
 	/* DAI Mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
@@ -545,6 +570,33 @@ static int sun8i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
 	u32 mode, val;
 	u8 offset;
 
+	/* DAI clock polarity */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_IF:
+		/* Invert both clocks */
+		val = SUN8I_I2S_FMT0_BCLK_POLARITY_INVERTED |
+		      SUN8I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/* Invert bit clock */
+		val = SUN8I_I2S_FMT0_BCLK_POLARITY_INVERTED;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		/* Invert frame clock */
+		val = SUN8I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		val = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
+			   SUN8I_I2S_FMT0_LRCLK_POLARITY_MASK |
+			   SUN8I_I2S_FMT0_BCLK_POLARITY_MASK,
+			   val);
+
 	/* DAI Mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
@@ -601,36 +653,8 @@ static int sun8i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
 static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	u32 bclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
-	u32 lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_NORMAL;
 	int ret;
 
-	/* DAI clock polarity */
-	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_IB_IF:
-		/* Invert both clocks */
-		bclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
-		lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
-		break;
-	case SND_SOC_DAIFMT_IB_NF:
-		/* Invert bit clock */
-		bclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
-		break;
-	case SND_SOC_DAIFMT_NB_IF:
-		/* Invert frame clock */
-		lrclk_polarity = SUN4I_I2S_FMT0_POLARITY_INVERTED;
-		break;
-	case SND_SOC_DAIFMT_NB_NF:
-		break;
-	default:
-		dev_err(dai->dev, "Unsupported clock polarity: %d\n",
-			fmt & SND_SOC_DAIFMT_INV_MASK);
-		return -EINVAL;
-	}
-
-	regmap_field_write(i2s->field_fmt_bclk, bclk_polarity);
-	regmap_field_write(i2s->field_fmt_lrclk, lrclk_polarity);
-
 	ret = i2s->variant->set_fmt(i2s, fmt);
 	if (ret) {
 		dev_err(dai->dev, "Unsupported format configuration\n");
@@ -984,8 +1008,6 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = {
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
-	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
-	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.bclk_dividers		= sun4i_i2s_bclk_div,
 	.num_bclk_dividers	= ARRAY_SIZE(sun4i_i2s_bclk_div),
 	.mclk_dividers		= sun4i_i2s_mclk_div,
@@ -1004,8 +1026,6 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
-	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
-	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.bclk_dividers		= sun4i_i2s_bclk_div,
 	.num_bclk_dividers	= ARRAY_SIZE(sun4i_i2s_bclk_div),
 	.mclk_dividers		= sun4i_i2s_mclk_div,
@@ -1024,8 +1044,6 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
-	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
-	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.bclk_dividers		= sun8i_i2s_clk_div,
 	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
 	.mclk_dividers		= sun8i_i2s_clk_div,
@@ -1045,8 +1063,6 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
-	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
-	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 19, 19),
 	.bclk_dividers		= sun8i_i2s_clk_div,
 	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
 	.mclk_dividers		= sun8i_i2s_clk_div,
@@ -1065,8 +1081,6 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
-	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
-	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
 	.bclk_dividers		= sun4i_i2s_bclk_div,
 	.num_bclk_dividers	= ARRAY_SIZE(sun4i_i2s_bclk_div),
 	.mclk_dividers		= sun4i_i2s_mclk_div,
@@ -1099,18 +1113,6 @@ static int sun4i_i2s_init_regmap_fields(struct device *dev,
 	if (IS_ERR(i2s->field_fmt_sr))
 		return PTR_ERR(i2s->field_fmt_sr);
 
-	i2s->field_fmt_bclk =
-			devm_regmap_field_alloc(dev, i2s->regmap,
-						i2s->variant->field_fmt_bclk);
-	if (IS_ERR(i2s->field_fmt_bclk))
-		return PTR_ERR(i2s->field_fmt_bclk);
-
-	i2s->field_fmt_lrclk =
-			devm_regmap_field_alloc(dev, i2s->regmap,
-						i2s->variant->field_fmt_lrclk);
-	if (IS_ERR(i2s->field_fmt_lrclk))
-		return PTR_ERR(i2s->field_fmt_lrclk);
-
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From dd657eae8164f7e4bafe8b875031a7c6c50646a9 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:20 +0200
Subject: ASoC: sun4i-i2s: Fix the LRCK polarity

The LRCK polarity "normal" polarity in the I2S/TDM specs and in the
Allwinner datasheet are not the same. In the case where the i2s controller
is being used as the LRCK master, it's pretty clear when looked at under a
scope.

Let's fix this, and add a comment to clear up as much the confusion as
possible.

Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3")
Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/e03fb6b2a916223070b9f18405b0ef117a452ff4.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 18 ++++++++++++------
 1 file changed, 12 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index e3eadfe38aaf..29b5eacd3abe 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -570,23 +570,29 @@ static int sun8i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
 	u32 mode, val;
 	u8 offset;
 
-	/* DAI clock polarity */
+	/*
+	 * DAI clock polarity
+	 *
+	 * The setup for LRCK contradicts the datasheet, but under a
+	 * scope it's clear that the LRCK polarity is reversed
+	 * compared to the expected polarity on the bus.
+	 */
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_IB_IF:
 		/* Invert both clocks */
-		val = SUN8I_I2S_FMT0_BCLK_POLARITY_INVERTED |
-		      SUN8I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
+		val = SUN8I_I2S_FMT0_BCLK_POLARITY_INVERTED;
 		break;
 	case SND_SOC_DAIFMT_IB_NF:
 		/* Invert bit clock */
-		val = SUN8I_I2S_FMT0_BCLK_POLARITY_INVERTED;
+		val = SUN8I_I2S_FMT0_BCLK_POLARITY_INVERTED |
+		      SUN8I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
 		break;
 	case SND_SOC_DAIFMT_NB_IF:
 		/* Invert frame clock */
-		val = SUN8I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
+		val = 0;
 		break;
 	case SND_SOC_DAIFMT_NB_NF:
-		val = 0;
+		val = SUN8I_I2S_FMT0_LRCLK_POLARITY_INVERTED;
 		break;
 	default:
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From 2e04fc4dbf50195262aa5a2ae6d35baa5b598cae Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:21 +0200
Subject: ASoC: sun4i-i2s: Fix WSS and SR fields for the A83t

The A83t has the same bit fields offsets than the A10 and A31, while this
was the first device with the new layout, fix that.

Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/d93f0943cc39d880750daf459a0eeab34c63518e.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 29b5eacd3abe..59d809df8d2a 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1048,8 +1048,8 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
-	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
-	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
 	.bclk_dividers		= sun8i_i2s_clk_div,
 	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
 	.mclk_dividers		= sun8i_i2s_clk_div,
-- 
cgit v1.2.3-59-g8ed1b


From bf943d527987c38f6fb11f9515e0cf2839286eb8 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:22 +0200
Subject: ASoC: sun4i-i2s: Fix MCLK Enable bit offset on A83t

The A83t, unlike previous SoCs, has the MCLK enable bit at the 8th bit of
the CLK_DIV register, unlike what is declared in the driver.

Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/43b07f8cd8e0e280c64ce61d57c307678c923e9b.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 59d809df8d2a..0fce3c476772 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1047,7 +1047,7 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
-	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
 	.bclk_dividers		= sun8i_i2s_clk_div,
-- 
cgit v1.2.3-59-g8ed1b


From 69e450e50ca6dde566f3ac3f2c329fb0492441ef Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:23 +0200
Subject: ASoC: sun4i-i2s: Fix the LRCK period on A83t

Unlike the previous SoCs, the A83t, like the newer ones, need the LRCK
bitfield to be set. Let's add it.

Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/6a0ee0bc1375bcb53840d3fb2d2f3d9732b8e57e.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 0fce3c476772..9468584f4eb0 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1047,6 +1047,7 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.has_fmt_set_lrck_period = true,
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
-- 
cgit v1.2.3-59-g8ed1b


From 3e9acd7ac6933cdc20c441bbf9a38ed9e42e1490 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:24 +0200
Subject: ASoC: sun4i-i2s: Remove duplicated quirks structure

The A83t and H3 have the same quirks, so it doesn't make sense to duplicate
the quirks structure.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/5ade5de27d23918c5ef30387c23aead951d5ad64.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 21 +--------------------
 1 file changed, 1 insertion(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9468584f4eb0..4c636f1cf7dc 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1062,25 +1062,6 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.set_fmt		= sun8i_i2s_set_soc_fmt,
 };
 
-static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
-	.has_reset		= true,
-	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
-	.sun4i_i2s_regmap	= &sun8i_i2s_regmap_config,
-	.has_fmt_set_lrck_period = true,
-	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
-	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
-	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
-	.bclk_dividers		= sun8i_i2s_clk_div,
-	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
-	.mclk_dividers		= sun8i_i2s_clk_div,
-	.num_mclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
-	.get_bclk_parent_rate	= sun8i_i2s_get_bclk_parent_rate,
-	.get_sr			= sun8i_i2s_get_sr_wss,
-	.get_wss		= sun8i_i2s_get_sr_wss,
-	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
-	.set_fmt		= sun8i_i2s_set_soc_fmt,
-};
-
 static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
@@ -1262,7 +1243,7 @@ static const struct of_device_id sun4i_i2s_match[] = {
 	},
 	{
 		.compatible = "allwinner,sun8i-h3-i2s",
-		.data = &sun8i_h3_i2s_quirks,
+		.data = &sun8i_a83t_i2s_quirks,
 	},
 	{
 		.compatible = "allwinner,sun50i-a64-codec-i2s",
-- 
cgit v1.2.3-59-g8ed1b


From 0083a507a78fdfa868acc0709408b59e72488a61 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:25 +0200
Subject: ASoC: sun4i-i2s: Pass the channels number as an argument

The channels number have been hardcoded to 2 so far, while the controller
supports more than that.

Remove the instance where it has been hardcoded to compute the BCLK
divider, and pass it through as an argument to ease further support of more
channels.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/48887cf7abfaab6597db233b24d7a088a913e48a.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 4c636f1cf7dc..6b172dfbc25d 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -238,10 +238,11 @@ static unsigned long sun8i_i2s_get_bclk_parent_rate(const struct sun4i_i2s *i2s)
 static int sun4i_i2s_get_bclk_div(struct sun4i_i2s *i2s,
 				  unsigned long parent_rate,
 				  unsigned int sampling_rate,
+				  unsigned int channels,
 				  unsigned int word_size)
 {
 	const struct sun4i_i2s_clk_div *dividers = i2s->variant->bclk_dividers;
-	int div = parent_rate / sampling_rate / word_size / 2;
+	int div = parent_rate / sampling_rate / word_size / channels;
 	int i;
 
 	for (i = 0; i < i2s->variant->num_bclk_dividers; i++) {
@@ -286,6 +287,7 @@ static bool sun4i_i2s_oversample_is_valid(unsigned int oversample)
 
 static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 				  unsigned int rate,
+				  unsigned int channels,
 				  unsigned int word_size)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
@@ -333,7 +335,7 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 
 	bclk_parent_rate = i2s->variant->get_bclk_parent_rate(i2s);
 	bclk_div = sun4i_i2s_get_bclk_div(i2s, bclk_parent_rate,
-					  rate, word_size);
+					  rate, channels, word_size);
 	if (bclk_div < 0) {
 		dev_err(dai->dev, "Unsupported BCLK divider: %d\n", bclk_div);
 		return -EINVAL;
@@ -488,7 +490,7 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 	regmap_field_write(i2s->field_fmt_sr, sr);
 
 	return sun4i_i2s_set_clk_rate(dai, params_rate(params),
-				      params_width(params));
+				      2, params_width(params));
 }
 
 static int sun4i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
-- 
cgit v1.2.3-59-g8ed1b


From bbf9a127abca4aac5cc75f882bc7efcc398e86ae Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:26 +0200
Subject: ASoC: sun4i-i2s: Support more channels

We've been limited to 2 channels in the driver while the controller
supports from 1 to 8 channels, in both capture and playback. let's remove
the hardcoded checks and numbers, and extend the range of channel numbers
we can use.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/27d9de5cd56f3a544851b8cd8af08bf836d19637.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 17 ++++++-----------
 1 file changed, 6 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 6b172dfbc25d..9e691baee1e8 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -400,9 +400,6 @@ static int sun4i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
 {
 	unsigned int channels = params_channels(params);
 
-	if (channels != 2)
-		return -EINVAL;
-
 	/* Map the channels for playback and capture */
 	regmap_write(i2s->regmap, SUN4I_I2S_TX_CHAN_MAP_REG, 0x76543210);
 	regmap_write(i2s->regmap, SUN4I_I2S_RX_CHAN_MAP_REG, 0x00003210);
@@ -423,9 +420,6 @@ static int sun8i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
 {
 	unsigned int channels = params_channels(params);
 
-	if (channels != 2)
-		return -EINVAL;
-
 	/* Map the channels for playback and capture */
 	regmap_write(i2s->regmap, SUN8I_I2S_TX_CHAN_MAP_REG, 0x76543210);
 	regmap_write(i2s->regmap, SUN8I_I2S_RX_CHAN_MAP_REG, 0x76543210);
@@ -458,6 +452,7 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 			       struct snd_soc_dai *dai)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int channels = params_channels(params);
 	int ret, sr, wss;
 	u32 width;
 
@@ -490,7 +485,7 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 	regmap_field_write(i2s->field_fmt_sr, sr);
 
 	return sun4i_i2s_set_clk_rate(dai, params_rate(params),
-				      2, params_width(params));
+				      channels, params_width(params));
 }
 
 static int sun4i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
@@ -814,15 +809,15 @@ static struct snd_soc_dai_driver sun4i_i2s_dai = {
 	.probe = sun4i_i2s_dai_probe,
 	.capture = {
 		.stream_name = "Capture",
-		.channels_min = 2,
-		.channels_max = 2,
+		.channels_min = 1,
+		.channels_max = 8,
 		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.playback = {
 		.stream_name = "Playback",
-		.channels_min = 2,
-		.channels_max = 2,
+		.channels_min = 1,
+		.channels_max = 8,
 		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-- 
cgit v1.2.3-59-g8ed1b


From 988b59467b2b14523a266957affbe9eca3e99fc9 Mon Sep 17 00:00:00 2001
From: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Date: Tue, 20 Aug 2019 19:24:09 +0300
Subject: ASoC: mchp-i2s-mcc: Fix unprepare of GCLK

If hw_free() gets called after hw_params(), GCLK remains prepared,
preventing further use of it. This patch fixes this by unpreparing the
clock in hw_free() or if hw_params() gets an error.

Fixes: 7e0cdf545a55 ("ASoC: mchp-i2s-mcc: add driver for I2SC Multi-Channel Controller")
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20190820162411.24836-2-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/mchp-i2s-mcc.c | 13 +++++++++++--
 1 file changed, 11 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 86495883ca3f..319f975586f1 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -670,8 +670,13 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	ret = regmap_write(dev->regmap, MCHP_I2SMCC_MRA, mra);
-	if (ret < 0)
+	if (ret < 0) {
+		if (dev->gclk_use) {
+			clk_unprepare(dev->gclk);
+			dev->gclk_use = 0;
+		}
 		return ret;
+	}
 	return regmap_write(dev->regmap, MCHP_I2SMCC_MRB, mrb);
 }
 
@@ -708,9 +713,13 @@ static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream,
 		regmap_write(dev->regmap, MCHP_I2SMCC_CR, MCHP_I2SMCC_CR_CKDIS);
 
 		if (dev->gclk_running) {
-			clk_disable_unprepare(dev->gclk);
+			clk_disable(dev->gclk);
 			dev->gclk_running = 0;
 		}
+		if (dev->gclk_use) {
+			clk_unprepare(dev->gclk);
+			dev->gclk_use = 0;
+		}
 	}
 
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From 0f6fc97501b790c971b11b52a654009d21c45238 Mon Sep 17 00:00:00 2001
From: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Date: Tue, 20 Aug 2019 19:24:10 +0300
Subject: ASoC: mchp-i2s-mcc: Wait for RX/TX RDY only if controller is running

Since hw_free() can be called multiple times and not just after a stop
trigger command, we should check whether the RX or TX ready interrupt was
truly enabled previously. For this, we assure that the condition of the
wait event is always true, except when RX/TX interrupts are enabled.

Fixes: 7e0cdf545a55 ("ASoC: mchp-i2s-mcc: add driver for I2SC Multi-Channel Controller")
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20190820162411.24836-3-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/mchp-i2s-mcc.c | 28 ++++++++++++++++------------
 1 file changed, 16 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 319f975586f1..ab7d5f98e759 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -691,22 +691,24 @@ static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream,
 		err = wait_event_interruptible_timeout(dev->wq_txrdy,
 						       dev->tx_rdy,
 						       msecs_to_jiffies(500));
+		if (err == 0) {
+			dev_warn_once(dev->dev,
+				      "Timeout waiting for Tx ready\n");
+			regmap_write(dev->regmap, MCHP_I2SMCC_IDRA,
+				     MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels));
+			dev->tx_rdy = 1;
+		}
 	} else {
 		err = wait_event_interruptible_timeout(dev->wq_rxrdy,
 						       dev->rx_rdy,
 						       msecs_to_jiffies(500));
-	}
-
-	if (err == 0) {
-		u32 idra;
-
-		dev_warn_once(dev->dev, "Timeout waiting for %s\n",
-			      is_playback ? "Tx ready" : "Rx ready");
-		if (is_playback)
-			idra = MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels);
-		else
-			idra = MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels);
-		regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, idra);
+		if (err == 0) {
+			dev_warn_once(dev->dev,
+				      "Timeout waiting for Rx ready\n");
+			regmap_write(dev->regmap, MCHP_I2SMCC_IDRA,
+				     MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels));
+			dev->rx_rdy = 1;
+		}
 	}
 
 	if (!mchp_i2s_mcc_is_running(dev)) {
@@ -818,6 +820,8 @@ static int mchp_i2s_mcc_dai_probe(struct snd_soc_dai *dai)
 
 	init_waitqueue_head(&dev->wq_txrdy);
 	init_waitqueue_head(&dev->wq_rxrdy);
+	dev->tx_rdy = 1;
+	dev->rx_rdy = 1;
 
 	snd_soc_dai_init_dma_data(dai, &dev->playback, &dev->capture);
 
-- 
cgit v1.2.3-59-g8ed1b


From c9cff337eab394c4dc8b128dde7308a1dd2e653a Mon Sep 17 00:00:00 2001
From: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Date: Tue, 20 Aug 2019 19:24:11 +0300
Subject: ASoC: mchp-i2s-mcc: Fix simultaneous capture and playback in master
 mode

This controller supports capture and playback running at the same time,
with the limitation that both capture and playback must be configured the
same way (sample rate, sample format, number of channels, etc). For this,
we have to assure that the configuration registers look the same when
capture and playback are initiated.
This patch fixes a bug in which the controller is in master mode and the
hw_params() callback fails for the second audio stream. The fail occurs
because the divisors are calculated after comparing the configuration
registers for capture and playback. The fix consists in calculating the
divisors before comparing the configuration registers. BCLK and LRC are
then configured and started only if the controller is not already running.

Fixes: 7e0cdf545a55 ("ASoC: mchp-i2s-mcc: add driver for I2SC Multi-Channel Controller")
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20190820162411.24836-4-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/mchp-i2s-mcc.c | 70 ++++++++++++++++++++++--------------------
 1 file changed, 37 insertions(+), 33 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 86495883ca3f..9a406144b18f 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -392,11 +392,11 @@ static int mchp_i2s_mcc_clk_get_rate_diff(struct clk *clk,
 }
 
 static int mchp_i2s_mcc_config_divs(struct mchp_i2s_mcc_dev *dev,
-				    unsigned int bclk, unsigned int *mra)
+				    unsigned int bclk, unsigned int *mra,
+				    unsigned long *best_rate)
 {
 	unsigned long clk_rate;
 	unsigned long lcm_rate;
-	unsigned long best_rate = 0;
 	unsigned long best_diff_rate = ~0;
 	unsigned int sysclk;
 	struct clk *best_clk = NULL;
@@ -423,7 +423,7 @@ static int mchp_i2s_mcc_config_divs(struct mchp_i2s_mcc_dev *dev,
 	     (clk_rate == bclk || clk_rate / (bclk * 2) <= GENMASK(5, 0));
 	     clk_rate += lcm_rate) {
 		ret = mchp_i2s_mcc_clk_get_rate_diff(dev->gclk, clk_rate,
-						     &best_clk, &best_rate,
+						     &best_clk, best_rate,
 						     &best_diff_rate);
 		if (ret) {
 			dev_err(dev->dev, "gclk error for rate %lu: %d",
@@ -437,7 +437,7 @@ static int mchp_i2s_mcc_config_divs(struct mchp_i2s_mcc_dev *dev,
 		}
 
 		ret = mchp_i2s_mcc_clk_get_rate_diff(dev->pclk, clk_rate,
-						     &best_clk, &best_rate,
+						     &best_clk, best_rate,
 						     &best_diff_rate);
 		if (ret) {
 			dev_err(dev->dev, "pclk error for rate %lu: %d",
@@ -459,33 +459,17 @@ static int mchp_i2s_mcc_config_divs(struct mchp_i2s_mcc_dev *dev,
 
 	dev_dbg(dev->dev, "source CLK is %s with rate %lu, diff %lu\n",
 		best_clk == dev->pclk ? "pclk" : "gclk",
-		best_rate, best_diff_rate);
-
-	/* set the rate */
-	ret = clk_set_rate(best_clk, best_rate);
-	if (ret) {
-		dev_err(dev->dev, "unable to set rate %lu to %s: %d\n",
-			best_rate, best_clk == dev->pclk ? "PCLK" : "GCLK",
-			ret);
-		return ret;
-	}
+		*best_rate, best_diff_rate);
 
 	/* Configure divisors */
 	if (dev->sysclk)
-		*mra |= MCHP_I2SMCC_MRA_IMCKDIV(best_rate / (2 * sysclk));
-	*mra |= MCHP_I2SMCC_MRA_ISCKDIV(best_rate / (2 * bclk));
+		*mra |= MCHP_I2SMCC_MRA_IMCKDIV(*best_rate / (2 * sysclk));
+	*mra |= MCHP_I2SMCC_MRA_ISCKDIV(*best_rate / (2 * bclk));
 
-	if (best_clk == dev->gclk) {
+	if (best_clk == dev->gclk)
 		*mra |= MCHP_I2SMCC_MRA_SRCCLK_GCLK;
-		ret = clk_prepare(dev->gclk);
-		if (ret < 0)
-			dev_err(dev->dev, "unable to prepare GCLK: %d\n", ret);
-		else
-			dev->gclk_use = 1;
-	} else {
+	else
 		*mra |= MCHP_I2SMCC_MRA_SRCCLK_PCLK;
-		dev->gclk_use = 0;
-	}
 
 	return 0;
 }
@@ -502,6 +486,7 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
 				  struct snd_pcm_hw_params *params,
 				  struct snd_soc_dai *dai)
 {
+	unsigned long rate = 0;
 	struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai);
 	u32 mra = 0;
 	u32 mrb = 0;
@@ -640,6 +625,17 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	if (set_divs) {
+		bclk_rate = frame_length * params_rate(params);
+		ret = mchp_i2s_mcc_config_divs(dev, bclk_rate, &mra,
+					       &rate);
+		if (ret) {
+			dev_err(dev->dev,
+				"unable to configure the divisors: %d\n", ret);
+			return ret;
+		}
+	}
+
 	/*
 	 * If we are already running, the wanted setup must be
 	 * the same with the one that's currently ongoing
@@ -656,19 +652,27 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
 		return 0;
 	}
 
-	/* Save the number of channels to know what interrupts to enable */
-	dev->channels = channels;
-
-	if (set_divs) {
-		bclk_rate = frame_length * params_rate(params);
-		ret = mchp_i2s_mcc_config_divs(dev, bclk_rate, &mra);
+	if (mra & MCHP_I2SMCC_MRA_SRCCLK_GCLK && !dev->gclk_use) {
+		/* set the rate */
+		ret = clk_set_rate(dev->gclk, rate);
 		if (ret) {
-			dev_err(dev->dev, "unable to configure the divisors: %d\n",
-				ret);
+			dev_err(dev->dev,
+				"unable to set rate %lu to GCLK: %d\n",
+				rate, ret);
+			return ret;
+		}
+
+		ret = clk_prepare(dev->gclk);
+		if (ret < 0) {
+			dev_err(dev->dev, "unable to prepare GCLK: %d\n", ret);
 			return ret;
 		}
+		dev->gclk_use = 1;
 	}
 
+	/* Save the number of channels to know what interrupts to enable */
+	dev->channels = channels;
+
 	ret = regmap_write(dev->regmap, MCHP_I2SMCC_MRA, mra);
 	if (ret < 0)
 		return ret;
-- 
cgit v1.2.3-59-g8ed1b


From 6eeb60be5ebb73b2e5911e26fb1aed02940b7d09 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 14 Aug 2019 11:29:10 +0300
Subject: ASoC: fsl_sai: Add support for imx8qm

SAI module on imx8qm features a register map similar with imx6 series
(it doesn't have VERID and PARAM registers at the beginning
of address spece).

Also, it has one FIFO which can help up to 64 * 32 bit samples.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20190814082911.665-2-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 4a346fcb5630..728307acab90 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1031,12 +1031,19 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = {
 	.reg_offset = 8,
 };
 
+static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = {
+	.use_imx_pcm = true,
+	.fifo_depth = 64,
+	.reg_offset = 0,
+};
+
 static const struct of_device_id fsl_sai_ids[] = {
 	{ .compatible = "fsl,vf610-sai", .data = &fsl_sai_vf610_data },
 	{ .compatible = "fsl,imx6sx-sai", .data = &fsl_sai_imx6sx_data },
 	{ .compatible = "fsl,imx6ul-sai", .data = &fsl_sai_imx6sx_data },
 	{ .compatible = "fsl,imx7ulp-sai", .data = &fsl_sai_imx7ulp_data },
 	{ .compatible = "fsl,imx8mq-sai", .data = &fsl_sai_imx8mq_data },
+	{ .compatible = "fsl,imx8qm-sai", .data = &fsl_sai_imx8qm_data },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(of, fsl_sai_ids);
-- 
cgit v1.2.3-59-g8ed1b


From b03bfaec1d52123d5d941488f71e06964535e471 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:04:54 +0900
Subject: ASoC: soc-core: merge snd_soc_initialize_card_lists()

snd_soc_initialize_card_lists() is doing card related
INIT_LIST_HEAD(), but, it is already doing at
snd_soc_register_card(). We don't need to do it separately.
This patch merges these.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/877e781ldq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 10 ----------
 sound/soc/soc-core.c | 13 ++++++++-----
 2 files changed, 8 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5c841c2ee814..f264c6509f00 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1220,16 +1220,6 @@ static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card)
 	return card->drvdata;
 }
 
-static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
-{
-	INIT_LIST_HEAD(&card->widgets);
-	INIT_LIST_HEAD(&card->paths);
-	INIT_LIST_HEAD(&card->dapm_list);
-	INIT_LIST_HEAD(&card->aux_comp_list);
-	INIT_LIST_HEAD(&card->component_dev_list);
-	INIT_LIST_HEAD(&card->list);
-}
-
 static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
 {
 	if (mc->reg == mc->rreg && mc->shift == mc->rshift)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b3f820fb53e6..d428491d51a7 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2370,15 +2370,18 @@ int snd_soc_register_card(struct snd_soc_card *card)
 
 	dev_set_drvdata(card->dev, card);
 
-	snd_soc_initialize_card_lists(card);
-
+	INIT_LIST_HEAD(&card->widgets);
+	INIT_LIST_HEAD(&card->paths);
+	INIT_LIST_HEAD(&card->dapm_list);
+	INIT_LIST_HEAD(&card->aux_comp_list);
+	INIT_LIST_HEAD(&card->component_dev_list);
+	INIT_LIST_HEAD(&card->list);
 	INIT_LIST_HEAD(&card->dai_link_list);
-
 	INIT_LIST_HEAD(&card->rtd_list);
-	card->num_rtd = 0;
-
 	INIT_LIST_HEAD(&card->dapm_dirty);
 	INIT_LIST_HEAD(&card->dobj_list);
+
+	card->num_rtd = 0;
 	card->instantiated = 0;
 	mutex_init(&card->mutex);
 	mutex_init(&card->dapm_mutex);
-- 
cgit v1.2.3-59-g8ed1b


From c26a8841119826badc8d358a4266880f83359f26 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:04:58 +0900
Subject: ASoC: soc-core: remove unneeded dai_link check from
 snd_soc_remove_dai_link()

snd_soc_remove_dai_link() has card connected dai_link check. but
	1) we need to call list_del() anyway,
	   because it is "remove" function,
	2) It is doing many thing for this card / dai_link already
	   before checking dai_link.

This patch removes poinless dai_link check

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/875zms1ldm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 9 +--------
 1 file changed, 1 insertion(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d428491d51a7..6df880be1622 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1182,8 +1182,6 @@ EXPORT_SYMBOL_GPL(snd_soc_add_dai_link);
 void snd_soc_remove_dai_link(struct snd_soc_card *card,
 			     struct snd_soc_dai_link *dai_link)
 {
-	struct snd_soc_dai_link *link, *_link;
-
 	if (dai_link->dobj.type
 	    && dai_link->dobj.type != SND_SOC_DOBJ_DAI_LINK) {
 		dev_err(card->dev, "Invalid dai link type %d\n",
@@ -1199,12 +1197,7 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 	if (dai_link->dobj.type && card->remove_dai_link)
 		card->remove_dai_link(card, dai_link);
 
-	for_each_card_links_safe(card, link, _link) {
-		if (link == dai_link) {
-			list_del(&link->list);
-			return;
-		}
-	}
+	list_del(&dai_link->list);
 }
 EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link);
 
-- 
cgit v1.2.3-59-g8ed1b


From 137befe19f310400a8b20fd8a4ce8c4141aafde0 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Mon, 19 Aug 2019 21:25:27 +0200
Subject: ASoC: sun4i-i2s: Add support for TDM slots

The i2s controller supports TDM, for up to 8 slots. Let's support the TDM
API.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/26392af30b3e7b31ee48d5b867d45be8675db046.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 40 ++++++++++++++++++++++++++++++++++------
 1 file changed, 34 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9e691baee1e8..8326b8cfa569 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -168,6 +168,8 @@ struct sun4i_i2s {
 	struct reset_control *rst;
 
 	unsigned int	mclk_freq;
+	unsigned int	slots;
+	unsigned int	slot_width;
 
 	struct snd_dmaengine_dai_dma_data	capture_dma_data;
 	struct snd_dmaengine_dai_dma_data	playback_dma_data;
@@ -287,7 +289,7 @@ static bool sun4i_i2s_oversample_is_valid(unsigned int oversample)
 
 static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 				  unsigned int rate,
-				  unsigned int channels,
+				  unsigned int slots,
 				  unsigned int word_size)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
@@ -335,7 +337,7 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 
 	bclk_parent_rate = i2s->variant->get_bclk_parent_rate(i2s);
 	bclk_div = sun4i_i2s_get_bclk_div(i2s, bclk_parent_rate,
-					  rate, channels, word_size);
+					  rate, slots, word_size);
 	if (bclk_div < 0) {
 		dev_err(dai->dev, "Unsupported BCLK divider: %d\n", bclk_div);
 		return -EINVAL;
@@ -419,6 +421,10 @@ static int sun8i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
 				  const struct snd_pcm_hw_params *params)
 {
 	unsigned int channels = params_channels(params);
+	unsigned int slots = channels;
+
+	if (i2s->slots)
+		slots = i2s->slots;
 
 	/* Map the channels for playback and capture */
 	regmap_write(i2s->regmap, SUN8I_I2S_TX_CHAN_MAP_REG, 0x76543210);
@@ -428,7 +434,6 @@ static int sun8i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
 	regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
 			   SUN4I_I2S_CHAN_SEL_MASK,
 			   SUN4I_I2S_CHAN_SEL(channels));
-
 	regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG,
 			   SUN4I_I2S_CHAN_SEL_MASK,
 			   SUN4I_I2S_CHAN_SEL(channels));
@@ -452,10 +457,18 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 			       struct snd_soc_dai *dai)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int word_size = params_width(params);
 	unsigned int channels = params_channels(params);
+	unsigned int slots = channels;
 	int ret, sr, wss;
 	u32 width;
 
+	if (i2s->slots)
+		slots = i2s->slots;
+
+	if (i2s->slot_width)
+		word_size = i2s->slot_width;
+
 	ret = i2s->variant->set_chan_cfg(i2s, params);
 	if (ret < 0) {
 		dev_err(dai->dev, "Invalid channel configuration\n");
@@ -477,15 +490,14 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (sr < 0)
 		return -EINVAL;
 
-	wss = i2s->variant->get_wss(i2s, params_width(params));
+	wss = i2s->variant->get_wss(i2s, word_size);
 	if (wss < 0)
 		return -EINVAL;
 
 	regmap_field_write(i2s->field_fmt_wss, wss);
 	regmap_field_write(i2s->field_fmt_sr, sr);
 
-	return sun4i_i2s_set_clk_rate(dai, params_rate(params),
-				      channels, params_width(params));
+	return sun4i_i2s_set_clk_rate(dai, params_rate(params), slots, word_size);
 }
 
 static int sun4i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
@@ -785,10 +797,26 @@ static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	return 0;
 }
 
+static int sun4i_i2s_set_tdm_slot(struct snd_soc_dai *dai,
+				  unsigned int tx_mask, unsigned int rx_mask,
+				  int slots, int slot_width)
+{
+	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	if (slots > 8)
+		return -EINVAL;
+
+	i2s->slots = slots;
+	i2s->slot_width = slot_width;
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
 	.hw_params	= sun4i_i2s_hw_params,
 	.set_fmt	= sun4i_i2s_set_fmt,
 	.set_sysclk	= sun4i_i2s_set_sysclk,
+	.set_tdm_slot	= sun4i_i2s_set_tdm_slot,
 	.trigger	= sun4i_i2s_trigger,
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 5389f4765789e4ecf9831bc968562befdd2f3bee Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Wed, 21 Aug 2019 15:06:53 +0200
Subject: ASoC: sun4i-i2s: Use the physical / slot width for the clocks

The clock dividers function has been using the word size to compute the
clock rate at which it's supposed to be running, but the proper formula
would be to use the physical width and / or slot width in TDM.

It doesn't make any difference at the moment since all the formats
supported have the same sample width and physical width, but it's not going
to last forever.

Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/41a359d9885f397e066816961e5e3236afcbe0a1.1566392800.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 12 +++++++-----
 1 file changed, 7 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 8326b8cfa569..cdc3fa60ff33 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -290,7 +290,7 @@ static bool sun4i_i2s_oversample_is_valid(unsigned int oversample)
 static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 				  unsigned int rate,
 				  unsigned int slots,
-				  unsigned int word_size)
+				  unsigned int slot_width)
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	unsigned int oversample_rate, clk_rate, bclk_parent_rate;
@@ -337,7 +337,7 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 
 	bclk_parent_rate = i2s->variant->get_bclk_parent_rate(i2s);
 	bclk_div = sun4i_i2s_get_bclk_div(i2s, bclk_parent_rate,
-					  rate, slots, word_size);
+					  rate, slots, slot_width);
 	if (bclk_div < 0) {
 		dev_err(dai->dev, "Unsupported BCLK divider: %d\n", bclk_div);
 		return -EINVAL;
@@ -458,6 +458,7 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	unsigned int word_size = params_width(params);
+	unsigned int slot_width = params_physical_width(params);
 	unsigned int channels = params_channels(params);
 	unsigned int slots = channels;
 	int ret, sr, wss;
@@ -467,7 +468,7 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 		slots = i2s->slots;
 
 	if (i2s->slot_width)
-		word_size = i2s->slot_width;
+		slot_width = i2s->slot_width;
 
 	ret = i2s->variant->set_chan_cfg(i2s, params);
 	if (ret < 0) {
@@ -490,14 +491,15 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (sr < 0)
 		return -EINVAL;
 
-	wss = i2s->variant->get_wss(i2s, word_size);
+	wss = i2s->variant->get_wss(i2s, slot_width);
 	if (wss < 0)
 		return -EINVAL;
 
 	regmap_field_write(i2s->field_fmt_wss, wss);
 	regmap_field_write(i2s->field_fmt_sr, sr);
 
-	return sun4i_i2s_set_clk_rate(dai, params_rate(params), slots, word_size);
+	return sun4i_i2s_set_clk_rate(dai, params_rate(params),
+				      slots, slot_width);
 }
 
 static int sun4i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
-- 
cgit v1.2.3-59-g8ed1b


From 9e8a93ac27d101e0ace024196a4bc3386568cc00 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Wed, 21 Aug 2019 15:06:54 +0200
Subject: ASoC: sun4i-i2s: Use the actual format width instead of an hardcoded
 one

The LRCK period field in the FMT0 register holds the number of LRCK period
for one channel in I2S mode.

This has been hardcoded to 32, while it really should be the physical width
of the format, which creates an improper clock when using a 16bit format,
with the i2s controller as LRCK master.

Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/f08a0c3605cd1d79752b38d704690190183f7865.1566392800.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index cdc3fa60ff33..9ef784b8867c 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -359,7 +359,7 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 	if (i2s->variant->has_fmt_set_lrck_period)
 		regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
 				   SUN8I_I2S_FMT0_LRCK_PERIOD_MASK,
-				   SUN8I_I2S_FMT0_LRCK_PERIOD(32));
+				   SUN8I_I2S_FMT0_LRCK_PERIOD(slot_width));
 
 	return 0;
 }
-- 
cgit v1.2.3-59-g8ed1b


From 84884c7ad5e8794aa19e48eaa8de93f4e1d26af9 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Wed, 21 Aug 2019 15:06:55 +0200
Subject: ASoC: sun4i-i2s: Replace call to params_width by local variable

The sun4i_i2s_hw_params function already has a variable holding the value
returned by params_width, so let's just use that variable instead of
calling params_width multiple times.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/f85a1c1e014080a4bbc3abd19bc8fdcb86f0981a.1566392800.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9ef784b8867c..69162af9fd65 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -487,7 +487,7 @@ static int sun4i_i2s_hw_params(struct snd_pcm_substream *substream,
 	}
 	i2s->playback_dma_data.addr_width = width;
 
-	sr = i2s->variant->get_sr(i2s, params_width(params));
+	sr = i2s->variant->get_sr(i2s, word_size);
 	if (sr < 0)
 		return -EINVAL;
 
-- 
cgit v1.2.3-59-g8ed1b


From 7ae7834ec446e5f7fed9bb990d16354853a206d0 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Wed, 21 Aug 2019 15:06:56 +0200
Subject: ASoC: sun4i-i2s: Add support for DSP formats

In addition to the I2S format, the controller also supports the DSP_*
formats.

This requires some extra care on the LRCK period calculation, since the
controller, with the PCM formats, require that the value set is no longer
the periods of LRCK for a single channel, but for all of them.

Let's add the code to deal with this, and support the DSP_A and DSP_B
formats.

Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/5562db1ac8759f12b1b87c3258223eed629ef771.1566392800.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 44 +++++++++++++++++++++++++++++++++++---------
 1 file changed, 35 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 69162af9fd65..57bf2a33753e 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -130,7 +130,6 @@ struct sun4i_i2s;
  * struct sun4i_i2s_quirks - Differences between SoC variants.
  *
  * @has_reset: SoC needs reset deasserted.
- * @has_fmt_set_lrck_period: SoC requires lrclk period to be set.
  * @reg_offset_txdata: offset of the tx fifo.
  * @sun4i_i2s_regmap: regmap config to use.
  * @field_clkdiv_mclk_en: regmap field to enable mclk output.
@@ -139,7 +138,6 @@ struct sun4i_i2s;
  */
 struct sun4i_i2s_quirks {
 	bool				has_reset;
-	bool				has_fmt_set_lrck_period;
 	unsigned int			reg_offset_txdata;	/* TX FIFO */
 	const struct regmap_config	*sun4i_i2s_regmap;
 
@@ -167,6 +165,7 @@ struct sun4i_i2s {
 	struct regmap	*regmap;
 	struct reset_control *rst;
 
+	unsigned int	format;
 	unsigned int	mclk_freq;
 	unsigned int	slots;
 	unsigned int	slot_width;
@@ -355,12 +354,6 @@ static int sun4i_i2s_set_clk_rate(struct snd_soc_dai *dai,
 
 	regmap_field_write(i2s->field_clkdiv_mclk_en, 1);
 
-	/* Set sync period */
-	if (i2s->variant->has_fmt_set_lrck_period)
-		regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
-				   SUN8I_I2S_FMT0_LRCK_PERIOD_MASK,
-				   SUN8I_I2S_FMT0_LRCK_PERIOD(slot_width));
-
 	return 0;
 }
 
@@ -422,6 +415,7 @@ static int sun8i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
 {
 	unsigned int channels = params_channels(params);
 	unsigned int slots = channels;
+	unsigned int lrck_period;
 
 	if (i2s->slots)
 		slots = i2s->slots;
@@ -445,6 +439,26 @@ static int sun8i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s,
 			   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK,
 			   SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(channels));
 
+	switch (i2s->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_RIGHT_J:
+		lrck_period = params_physical_width(params) * slots;
+		break;
+
+	case SND_SOC_DAIFMT_I2S:
+		lrck_period = params_physical_width(params);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_FMT0_REG,
+			   SUN8I_I2S_FMT0_LRCK_PERIOD_MASK,
+			   SUN8I_I2S_FMT0_LRCK_PERIOD(lrck_period));
+
 	regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG,
 			   SUN8I_I2S_TX_CHAN_EN_MASK,
 			   SUN8I_I2S_TX_CHAN_EN(channels));
@@ -616,6 +630,16 @@ static int sun8i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s,
 
 	/* DAI Mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		mode = SUN8I_I2S_CTRL_MODE_PCM;
+		offset = 1;
+		break;
+
+	case SND_SOC_DAIFMT_DSP_B:
+		mode = SUN8I_I2S_CTRL_MODE_PCM;
+		offset = 0;
+		break;
+
 	case SND_SOC_DAIFMT_I2S:
 		mode = SUN8I_I2S_CTRL_MODE_LEFT;
 		offset = 1;
@@ -684,6 +708,9 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 			   SUN4I_I2S_FIFO_CTRL_RX_MODE_MASK,
 			   SUN4I_I2S_FIFO_CTRL_TX_MODE(1) |
 			   SUN4I_I2S_FIFO_CTRL_RX_MODE(1));
+
+	i2s->format = fmt;
+
 	return 0;
 }
 
@@ -1074,7 +1101,6 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
-	.has_fmt_set_lrck_period = true,
 	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
 	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
 	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
-- 
cgit v1.2.3-59-g8ed1b


From 2722b535337b35a9c0be50de687cccda4ab9468d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Aug 2019 09:55:37 +0200
Subject: ALSA: hda/realtek - Check beep whitelist before assigning in all
 codecs

Some Realtek codec parsers didn't check the availability of PC beep.
Add has_cdefine_beep() check appropriately.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 9 ++++++---
 1 file changed, 6 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b6ae452a1d8f..19831521bd95 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2848,7 +2848,8 @@ static int patch_alc268(struct hda_codec *codec)
 		return err;
 
 	spec = codec->spec;
-	spec->gen.beep_nid = 0x01;
+	if (has_cdefine_beep(codec))
+		spec->gen.beep_nid = 0x01;
 
 	spec->shutup = alc_eapd_shutup;
 
@@ -8102,7 +8103,8 @@ static int patch_alc861(struct hda_codec *codec)
 		return err;
 
 	spec = codec->spec;
-	spec->gen.beep_nid = 0x23;
+	if (has_cdefine_beep(codec))
+		spec->gen.beep_nid = 0x23;
 
 #ifdef CONFIG_PM
 	spec->power_hook = alc_power_eapd;
@@ -8203,7 +8205,8 @@ static int patch_alc861vd(struct hda_codec *codec)
 		return err;
 
 	spec = codec->spec;
-	spec->gen.beep_nid = 0x23;
+	if (has_cdefine_beep(codec))
+		spec->gen.beep_nid = 0x23;
 
 	spec->shutup = alc_eapd_shutup;
 
-- 
cgit v1.2.3-59-g8ed1b


From 051c78af14fcd74a22b5af45548ad9d588247cc7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Aug 2019 09:58:07 +0200
Subject: ALSA: hda/realtek - Blacklist PC beep for Lenovo ThinkCentre M73/93

Lenovo ThinkCentre M73 and M93 don't seem to have a proper beep
although the driver tries to probe and set up blindly.
Blacklist these machines for suppressing the beep creation.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204635
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 19831521bd95..0585484a469a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1065,6 +1065,9 @@ static const struct snd_pci_quirk beep_white_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
 	SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1),
 	SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
+	/* blacklist -- no beep available */
+	SND_PCI_QUIRK(0x17aa, 0x309e, "Lenovo ThinkCentre M73", 0),
+	SND_PCI_QUIRK(0x17aa, 0x30a3, "Lenovo ThinkCentre M93", 0),
 	{}
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 57f8770620e9b51c61089751f0b5ad3dbe376ff2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Aug 2019 17:17:09 +0200
Subject: ALSA: usb-audio: More validations of descriptor units

Introduce a new helper to validate each audio descriptor unit before
and check the unit before actually accessing it.  This should harden
against the OOB access cases with malformed descriptors that have been
recently frequently reported by fuzzers.

The existing descriptor checks are still kept although they become
superfluous after this patch.  They'll be cleaned up eventually
later.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/Makefile   |   3 +-
 sound/usb/helper.h   |   4 +
 sound/usb/mixer.c    |  10 ++
 sound/usb/power.c    |   2 +
 sound/usb/quirks.c   |   3 +
 sound/usb/stream.c   |  25 ++--
 sound/usb/validate.c | 332 +++++++++++++++++++++++++++++++++++++++++++++++++++
 7 files changed, 366 insertions(+), 13 deletions(-)
 create mode 100644 sound/usb/validate.c

(limited to 'sound')

diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index e1ce257ab705..d27a21b0ff9c 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -16,7 +16,8 @@ snd-usb-audio-objs := 	card.o \
 			power.o \
 			proc.o \
 			quirks.o \
-			stream.o
+			stream.o \
+			validate.o
 
 snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o
 
diff --git a/sound/usb/helper.h b/sound/usb/helper.h
index 6afb70156ec4..5e8a18b4e7b9 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -31,4 +31,8 @@ static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip)
 	return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber;
 }
 
+/* in validate.c */
+bool snd_usb_validate_audio_desc(void *p, int protocol);
+bool snd_usb_validate_midi_desc(void *p);
+
 #endif /* __USBAUDIO_HELPER_H */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index eceab19766db..a1093fb9bf09 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -785,6 +785,8 @@ static int __check_input_term(struct mixer_build *state, int id,
 		p1 = find_audio_control_unit(state, id);
 		if (!p1)
 			break;
+		if (!snd_usb_validate_audio_desc(p1, protocol))
+			break; /* bad descriptor */
 
 		hdr = p1;
 		term->id = id;
@@ -2775,6 +2777,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 		return -EINVAL;
 	}
 
+	if (!snd_usb_validate_audio_desc(p1, protocol)) {
+		usb_audio_dbg(state->chip, "invalid unit %d\n", unitid);
+		return 0; /* skip invalid unit */
+	}
+
 	if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
 		switch (p1[2]) {
 		case UAC_INPUT_TERMINAL:
@@ -3145,6 +3152,9 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 	while ((p = snd_usb_find_csint_desc(mixer->hostif->extra,
 					    mixer->hostif->extralen,
 					    p, UAC_OUTPUT_TERMINAL)) != NULL) {
+		if (!snd_usb_validate_audio_desc(p, mixer->protocol))
+			continue; /* skip invalid descriptor */
+
 		if (mixer->protocol == UAC_VERSION_1) {
 			struct uac1_output_terminal_descriptor *desc = p;
 
diff --git a/sound/usb/power.c b/sound/usb/power.c
index bd303a1ba1b7..606a2cb23eab 100644
--- a/sound/usb/power.c
+++ b/sound/usb/power.c
@@ -31,6 +31,8 @@ snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface,
 		struct uac3_power_domain_descriptor *pd_desc = p;
 		int i;
 
+		if (!snd_usb_validate_audio_desc(p, UAC_VERSION_3))
+			continue;
 		for (i = 0; i < pd_desc->bNrEntities; i++) {
 			if (pd_desc->baEntityID[i] == id) {
 				pd->pd_id = pd_desc->bPowerDomainID;
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 78858918cbc1..7e9735aa7ac9 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -248,6 +248,9 @@ static int create_yamaha_midi_quirk(struct snd_usb_audio *chip,
 					NULL, USB_MS_MIDI_OUT_JACK);
 	if (!injd && !outjd)
 		return -ENODEV;
+	if (!snd_usb_validate_midi_desc(injd) ||
+	    !snd_usb_validate_midi_desc(outjd))
+		return -ENODEV;
 	if (injd && (injd->bLength < 5 ||
 		     (injd->bJackType != USB_MS_EMBEDDED &&
 		      injd->bJackType != USB_MS_EXTERNAL)))
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index e852c7fd6109..a0649c8ae460 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -627,16 +627,14 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
  */
 static void *
 snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-				       int terminal_id, bool uac23)
+				       int terminal_id, int protocol)
 {
 	struct uac2_input_terminal_descriptor *term = NULL;
-	size_t minlen = uac23 ? sizeof(struct uac2_input_terminal_descriptor) :
-		sizeof(struct uac_input_terminal_descriptor);
 
 	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
 					       ctrl_iface->extralen,
 					       term, UAC_INPUT_TERMINAL))) {
-		if (term->bLength < minlen)
+		if (!snd_usb_validate_audio_desc(term, protocol))
 			continue;
 		if (term->bTerminalID == terminal_id)
 			return term;
@@ -647,7 +645,7 @@ snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
 
 static void *
 snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
-					int terminal_id)
+					int terminal_id, int protocol)
 {
 	/* OK to use with both UAC2 and UAC3 */
 	struct uac2_output_terminal_descriptor *term = NULL;
@@ -655,8 +653,9 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
 	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
 					       ctrl_iface->extralen,
 					       term, UAC_OUTPUT_TERMINAL))) {
-		if (term->bLength >= sizeof(*term) &&
-		    term->bTerminalID == terminal_id)
+		if (!snd_usb_validate_audio_desc(term, protocol))
+			continue;
+		if (term->bTerminalID == terminal_id)
 			return term;
 	}
 
@@ -731,7 +730,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
 
 		iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
 							       as->bTerminalLink,
-							       false);
+							       protocol);
 		if (iterm) {
 			num_channels = iterm->bNrChannels;
 			chconfig = le16_to_cpu(iterm->wChannelConfig);
@@ -767,7 +766,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
 		 */
 		input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
 								    as->bTerminalLink,
-								    true);
+								    protocol);
 		if (input_term) {
 			clock = input_term->bCSourceID;
 			if (!chconfig && (num_channels == input_term->bNrChannels))
@@ -776,7 +775,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
 		}
 
 		output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
-								      as->bTerminalLink);
+								      as->bTerminalLink,
+								      protocol);
 		if (output_term) {
 			clock = output_term->bCSourceID;
 			goto found_clock;
@@ -1002,14 +1002,15 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip,
 	 */
 	input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
 							    as->bTerminalLink,
-							    true);
+							    UAC_VERSION_3);
 	if (input_term) {
 		clock = input_term->bCSourceID;
 		goto found_clock;
 	}
 
 	output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
-							     as->bTerminalLink);
+							      as->bTerminalLink,
+							      UAC_VERSION_3);
 	if (output_term) {
 		clock = output_term->bCSourceID;
 		goto found_clock;
diff --git a/sound/usb/validate.c b/sound/usb/validate.c
new file mode 100644
index 000000000000..3c8f73a0eb12
--- /dev/null
+++ b/sound/usb/validate.c
@@ -0,0 +1,332 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+//
+// Validation of USB-audio class descriptors
+//
+
+#include <linux/init.h>
+#include <linux/usb.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/audio-v2.h>
+#include <linux/usb/audio-v3.h>
+#include <linux/usb/midi.h>
+#include "usbaudio.h"
+#include "helper.h"
+
+struct usb_desc_validator {
+	unsigned char protocol;
+	unsigned char type;
+	bool (*func)(const void *p, const struct usb_desc_validator *v);
+	size_t size;
+};
+
+#define UAC_VERSION_ALL		(unsigned char)(-1)
+
+/* UAC1 only */
+static bool validate_uac1_header(const void *p,
+				 const struct usb_desc_validator *v)
+{
+	const struct uac1_ac_header_descriptor *d = p;
+
+	return d->bLength >= sizeof(*d) &&
+		d->bLength >= sizeof(*d) + d->bInCollection;
+}
+
+/* for mixer unit; covering all UACs */
+static bool validate_mixer_unit(const void *p,
+				const struct usb_desc_validator *v)
+{
+	const struct uac_mixer_unit_descriptor *d = p;
+	size_t len;
+
+	if (d->bLength < sizeof(*d) || !d->bNrInPins)
+		return false;
+	len = sizeof(*d) + d->bNrInPins;
+	/* We can't determine the bitmap size only from this unit descriptor,
+	 * so just check with the remaining length.
+	 * The actual bitmap is checked at mixer unit parser.
+	 */
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		len += 2 + 1; /* wChannelConfig, iChannelNames */
+		/* bmControls[n*m] */
+		len += 1; /* iMixer */
+		break;
+	case UAC_VERSION_2:
+		len += 4 + 1; /* bmChannelConfig, iChannelNames */
+		/* bmMixerControls[n*m] */
+		len += 1 + 1; /* bmControls, iMixer */
+		break;
+	case UAC_VERSION_3:
+		len += 2; /* wClusterDescrID */
+		/* bmMixerControls[n*m] */
+		break;
+	}
+	return d->bLength >= len;
+}
+
+/* both for processing and extension units; covering all UACs */
+static bool validate_processing_unit(const void *p,
+				     const struct usb_desc_validator *v)
+{
+	const struct uac_processing_unit_descriptor *d = p;
+	const unsigned char *hdr = p;
+	size_t len, m;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	len = d->bLength < sizeof(*d) + d->bNrInPins;
+	if (d->bLength < len)
+		return false;
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		/* bNrChannels, wChannelConfig, iChannelNames, bControlSize */
+		len += 1 + 2 + 1 + 1;
+		if (d->bLength < len) /* bControlSize */
+			return false;
+		m = hdr[len];
+		len += 1 + m + 1; /* bControlSize, bmControls, iProcessing */
+		break;
+	case UAC_VERSION_2:
+		/* bNrChannels, bmChannelConfig, iChannelNames */
+		len += 1 + 4 + 1;
+		if (v->type == UAC2_PROCESSING_UNIT_V2)
+			len += 2; /* bmControls -- 2 bytes for PU */
+		else
+			len += 1; /* bmControls -- 1 byte for EU */
+		len += 1; /* iProcessing */
+		break;
+	case UAC_VERSION_3:
+		/* wProcessingDescrStr, bmControls */
+		len += 2 + 4;
+		break;
+	}
+	if (d->bLength < len)
+		return false;
+
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		if (v->type == UAC1_EXTENSION_UNIT)
+			return true; /* OK */
+		switch (d->wProcessType) {
+		case UAC_PROCESS_UP_DOWNMIX:
+		case UAC_PROCESS_DOLBY_PROLOGIC:
+			if (d->bLength < len + 1) /* bNrModes */
+				return false;
+			m = hdr[len];
+			len += 1 + m * 2; /* bNrModes, waModes(n) */
+			break;
+		default:
+			break;
+		}
+		break;
+	case UAC_VERSION_2:
+		if (v->type == UAC2_EXTENSION_UNIT_V2)
+			return true; /* OK */
+		switch (d->wProcessType) {
+		case UAC2_PROCESS_UP_DOWNMIX:
+		case UAC2_PROCESS_DOLBY_PROLOCIC: /* SiC! */
+			if (d->bLength < len + 1) /* bNrModes */
+				return false;
+			m = hdr[len];
+			len += 1 + m * 4; /* bNrModes, daModes(n) */
+			break;
+		default:
+			break;
+		}
+		break;
+	case UAC_VERSION_3:
+		if (v->type == UAC3_EXTENSION_UNIT) {
+			len += 2; /* wClusterDescrID */
+			break;
+		}
+		switch (d->wProcessType) {
+		case UAC3_PROCESS_UP_DOWNMIX:
+			if (d->bLength < len + 1) /* bNrModes */
+				return false;
+			m = hdr[len];
+			len += 1 + m * 2; /* bNrModes, waClusterDescrID(n) */
+			break;
+		case UAC3_PROCESS_MULTI_FUNCTION:
+			len += 2 + 4; /* wClusterDescrID, bmAlgorighms */
+			break;
+		default:
+			break;
+		}
+		break;
+	}
+	if (d->bLength < len)
+		return false;
+
+	return true;
+}
+
+/* both for selector and clock selector units; covering all UACs */
+static bool validate_selector_unit(const void *p,
+				   const struct usb_desc_validator *v)
+{
+	const struct uac_selector_unit_descriptor *d = p;
+	size_t len;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	len = sizeof(*d) + d->bNrInPins;
+	switch (v->protocol) {
+	case UAC_VERSION_1:
+	default:
+		len += 1; /* iSelector */
+		break;
+	case UAC_VERSION_2:
+		len += 1 + 1; /* bmControls, iSelector */
+		break;
+	case UAC_VERSION_3:
+		len += 4 + 2; /* bmControls, wSelectorDescrStr */
+		break;
+	}
+	return d->bLength >= len;
+}
+
+static bool validate_uac1_feature_unit(const void *p,
+				       const struct usb_desc_validator *v)
+{
+	const struct uac_feature_unit_descriptor *d = p;
+
+	if (d->bLength < sizeof(*d) || !d->bControlSize)
+		return false;
+	/* at least bmaControls(0) for master channel + iFeature */
+	return d->bLength >= sizeof(*d) + d->bControlSize + 1;
+}
+
+static bool validate_uac2_feature_unit(const void *p,
+				       const struct usb_desc_validator *v)
+{
+	const struct uac2_feature_unit_descriptor *d = p;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	/* at least bmaControls(0) for master channel + iFeature */
+	return d->bLength >= sizeof(*d) + 4 + 1;
+}
+
+static bool validate_uac3_feature_unit(const void *p,
+				       const struct usb_desc_validator *v)
+{
+	const struct uac3_feature_unit_descriptor *d = p;
+
+	if (d->bLength < sizeof(*d))
+		return false;
+	/* at least bmaControls(0) for master channel + wFeatureDescrStr */
+	return d->bLength >= sizeof(*d) + 4 + 2;
+}
+
+static bool validate_midi_out_jack(const void *p,
+				   const struct usb_desc_validator *v)
+{
+	const struct usb_midi_out_jack_descriptor *d = p;
+
+	return d->bLength >= sizeof(*d) &&
+		d->bLength >= sizeof(*d) + d->bNrInputPins * 2;
+}
+
+#define FIXED(p, t, s) { .protocol = (p), .type = (t), .size = sizeof(s) }
+#define FUNC(p, t, f) { .protocol = (p), .type = (t), .func = (f) }
+
+static struct usb_desc_validator audio_validators[] = {
+	/* UAC1 */
+	FUNC(UAC_VERSION_1, UAC_HEADER, validate_uac1_header),
+	FIXED(UAC_VERSION_1, UAC_INPUT_TERMINAL,
+	      struct uac_input_terminal_descriptor),
+	FIXED(UAC_VERSION_1, UAC_OUTPUT_TERMINAL,
+	      struct uac1_output_terminal_descriptor),
+	FUNC(UAC_VERSION_1, UAC_MIXER_UNIT, validate_mixer_unit),
+	FUNC(UAC_VERSION_1, UAC_SELECTOR_UNIT, validate_selector_unit),
+	FUNC(UAC_VERSION_1, UAC_FEATURE_UNIT, validate_uac1_feature_unit),
+	FUNC(UAC_VERSION_1, UAC1_PROCESSING_UNIT, validate_processing_unit),
+	FUNC(UAC_VERSION_1, UAC1_EXTENSION_UNIT, validate_processing_unit),
+
+	/* UAC2 */
+	FIXED(UAC_VERSION_2, UAC_HEADER, struct uac2_ac_header_descriptor),
+	FIXED(UAC_VERSION_2, UAC_INPUT_TERMINAL,
+	      struct uac2_input_terminal_descriptor),
+	FIXED(UAC_VERSION_2, UAC_OUTPUT_TERMINAL,
+	      struct uac2_output_terminal_descriptor),
+	FUNC(UAC_VERSION_2, UAC_MIXER_UNIT, validate_mixer_unit),
+	FUNC(UAC_VERSION_2, UAC_SELECTOR_UNIT, validate_selector_unit),
+	FUNC(UAC_VERSION_2, UAC_FEATURE_UNIT, validate_uac2_feature_unit),
+	/* UAC_VERSION_2, UAC2_EFFECT_UNIT: not implemented yet */
+	FUNC(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2, validate_processing_unit),
+	FUNC(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2, validate_processing_unit),
+	FIXED(UAC_VERSION_2, UAC2_CLOCK_SOURCE,
+	      struct uac_clock_source_descriptor),
+	FUNC(UAC_VERSION_2, UAC2_CLOCK_SELECTOR, validate_selector_unit),
+	FIXED(UAC_VERSION_2, UAC2_CLOCK_MULTIPLIER,
+	      struct uac_clock_multiplier_descriptor),
+	/* UAC_VERSION_2, UAC2_SAMPLE_RATE_CONVERTER: not implemented yet */
+
+	/* UAC3 */
+	FIXED(UAC_VERSION_2, UAC_HEADER, struct uac3_ac_header_descriptor),
+	FIXED(UAC_VERSION_3, UAC_INPUT_TERMINAL,
+	      struct uac3_input_terminal_descriptor),
+	FIXED(UAC_VERSION_3, UAC_OUTPUT_TERMINAL,
+	      struct uac3_output_terminal_descriptor),
+	/* UAC_VERSION_3, UAC3_EXTENDED_TERMINAL: not implemented yet */
+	FUNC(UAC_VERSION_3, UAC3_MIXER_UNIT, validate_mixer_unit),
+	FUNC(UAC_VERSION_3, UAC3_SELECTOR_UNIT, validate_selector_unit),
+	FUNC(UAC_VERSION_3, UAC_FEATURE_UNIT, validate_uac3_feature_unit),
+	/*  UAC_VERSION_3, UAC3_EFFECT_UNIT: not implemented yet */
+	FUNC(UAC_VERSION_3, UAC3_PROCESSING_UNIT, validate_processing_unit),
+	FUNC(UAC_VERSION_3, UAC3_EXTENSION_UNIT, validate_processing_unit),
+	FIXED(UAC_VERSION_3, UAC3_CLOCK_SOURCE,
+	      struct uac3_clock_source_descriptor),
+	FUNC(UAC_VERSION_3, UAC3_CLOCK_SELECTOR, validate_selector_unit),
+	FIXED(UAC_VERSION_3, UAC3_CLOCK_MULTIPLIER,
+	      struct uac3_clock_multiplier_descriptor),
+	/* UAC_VERSION_3, UAC3_SAMPLE_RATE_CONVERTER: not implemented yet */
+	/* UAC_VERSION_3, UAC3_CONNECTORS: not implemented yet */
+	{ } /* terminator */
+};
+
+static struct usb_desc_validator midi_validators[] = {
+	FIXED(UAC_VERSION_ALL, USB_MS_HEADER,
+	      struct usb_ms_header_descriptor),
+	FIXED(UAC_VERSION_ALL, USB_MS_MIDI_IN_JACK,
+	      struct usb_midi_in_jack_descriptor),
+	FUNC(UAC_VERSION_ALL, USB_MS_MIDI_OUT_JACK,
+	     validate_midi_out_jack),
+	{ } /* terminator */
+};
+
+
+/* Validate the given unit descriptor, return true if it's OK */
+static bool validate_desc(unsigned char *hdr, int protocol,
+			  const struct usb_desc_validator *v)
+{
+	if (hdr[1] != USB_DT_CS_INTERFACE)
+		return true; /* don't care */
+
+	for (; v->type; v++) {
+		if (v->type == hdr[2] &&
+		    (v->protocol == UAC_VERSION_ALL ||
+		     v->protocol == protocol)) {
+			if (v->func)
+				return v->func(hdr, v);
+			/* check for the fixed size */
+			return hdr[0] >= v->size;
+		}
+	}
+
+	return true; /* not matching, skip validation */
+}
+
+bool snd_usb_validate_audio_desc(void *p, int protocol)
+{
+	return validate_desc(p, protocol, audio_validators);
+}
+
+bool snd_usb_validate_midi_desc(void *p)
+{
+	return validate_desc(p, UAC_VERSION_1, midi_validators);
+}
+
-- 
cgit v1.2.3-59-g8ed1b


From 68e9fde245591d18200f8a9054cac22339437adb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 15 Aug 2019 16:30:39 +0200
Subject: ALSA: usb-audio: Simplify parse_audio_unit()

Minor code refactoring by combining the UAC version and the type in
the switch-case flow, so that we reduce the indentation and
redundancy.  One good bonus is that the duplicated definition of the
same type value (e.g. UAC2_EFFECT_UNIT) can be handled more cleanly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 95 +++++++++++++++++++++++--------------------------------
 1 file changed, 39 insertions(+), 56 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index a1093fb9bf09..34def5911af2 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2782,62 +2782,45 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 		return 0; /* skip invalid unit */
 	}
 
-	if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
-		switch (p1[2]) {
-		case UAC_INPUT_TERMINAL:
-			return parse_audio_input_terminal(state, unitid, p1);
-		case UAC_MIXER_UNIT:
-			return parse_audio_mixer_unit(state, unitid, p1);
-		case UAC2_CLOCK_SOURCE:
-			return parse_clock_source_unit(state, unitid, p1);
-		case UAC_SELECTOR_UNIT:
-		case UAC2_CLOCK_SELECTOR:
-			return parse_audio_selector_unit(state, unitid, p1);
-		case UAC_FEATURE_UNIT:
-			return parse_audio_feature_unit(state, unitid, p1);
-		case UAC1_PROCESSING_UNIT:
-		/*   UAC2_EFFECT_UNIT has the same value */
-			if (protocol == UAC_VERSION_1)
-				return parse_audio_processing_unit(state, unitid, p1);
-			else
-				return 0; /* FIXME - effect units not implemented yet */
-		case UAC1_EXTENSION_UNIT:
-		/*   UAC2_PROCESSING_UNIT_V2 has the same value */
-			if (protocol == UAC_VERSION_1)
-				return parse_audio_extension_unit(state, unitid, p1);
-			else /* UAC_VERSION_2 */
-				return parse_audio_processing_unit(state, unitid, p1);
-		case UAC2_EXTENSION_UNIT_V2:
-			return parse_audio_extension_unit(state, unitid, p1);
-		default:
-			usb_audio_err(state->chip,
-				"unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
-			return -EINVAL;
-		}
-	} else { /* UAC_VERSION_3 */
-		switch (p1[2]) {
-		case UAC_INPUT_TERMINAL:
-			return parse_audio_input_terminal(state, unitid, p1);
-		case UAC3_MIXER_UNIT:
-			return parse_audio_mixer_unit(state, unitid, p1);
-		case UAC3_CLOCK_SOURCE:
-			return parse_clock_source_unit(state, unitid, p1);
-		case UAC3_SELECTOR_UNIT:
-		case UAC3_CLOCK_SELECTOR:
-			return parse_audio_selector_unit(state, unitid, p1);
-		case UAC3_FEATURE_UNIT:
-			return parse_audio_feature_unit(state, unitid, p1);
-		case UAC3_EFFECT_UNIT:
-			return 0; /* FIXME - effect units not implemented yet */
-		case UAC3_PROCESSING_UNIT:
-			return parse_audio_processing_unit(state, unitid, p1);
-		case UAC3_EXTENSION_UNIT:
-			return parse_audio_extension_unit(state, unitid, p1);
-		default:
-			usb_audio_err(state->chip,
-				"unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
-			return -EINVAL;
-		}
+#define PTYPE(a, b)	((a) << 8 | (b))
+	switch (PTYPE(protocol, p1[2])) {
+	case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL):
+	case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL):
+	case PTYPE(UAC_VERSION_3, UAC_INPUT_TERMINAL):
+		return parse_audio_input_terminal(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC_MIXER_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC_MIXER_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_MIXER_UNIT):
+		return parse_audio_mixer_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SOURCE):
+	case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SOURCE):
+		return parse_clock_source_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC_SELECTOR_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC_SELECTOR_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_SELECTOR_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SELECTOR):
+	case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SELECTOR):
+		return parse_audio_selector_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC_FEATURE_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC_FEATURE_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_FEATURE_UNIT):
+		return parse_audio_feature_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC1_PROCESSING_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2):
+	case PTYPE(UAC_VERSION_3, UAC3_PROCESSING_UNIT):
+		return parse_audio_processing_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT):
+	case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2):
+	case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT):
+		return parse_audio_extension_unit(state, unitid, p1);
+	case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT):
+	case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT):
+		return 0; /* FIXME - effect units not implemented yet */
+	default:
+		usb_audio_err(state->chip,
+			      "unit %u: unexpected type 0x%02x\n",
+			      unitid, p1[2]);
+		return -EINVAL;
 	}
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 52c3e317a857091fd746e15179a637f32be4d337 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Aug 2019 08:23:10 +0200
Subject: ALSA: usb-audio: Unify the release of usb_mixer_elem_info objects

Instead of the direct kfree() calls, introduce a new local helper to
release the usb_mixer_elem_info object.  This will be extended to do
more than a single kfree() in the later patches.

Also, use the standard goto instead of multiple calls in
parse_audio_selector_unit() error paths.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 48 ++++++++++++++++++++++++++++--------------------
 1 file changed, 28 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 34def5911af2..277660fd6e0a 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1026,10 +1026,15 @@ static struct usb_feature_control_info audio_feature_info[] = {
 	{ UAC2_FU_PHASE_INVERTER,	 "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 },
 };
 
+static void usb_mixer_elem_info_free(struct usb_mixer_elem_info *cval)
+{
+	kfree(cval);
+}
+
 /* private_free callback */
 void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl)
 {
-	kfree(kctl->private_data);
+	usb_mixer_elem_info_free(kctl->private_data);
 	kctl->private_data = NULL;
 }
 
@@ -1552,7 +1557,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
 
 	ctl_info = get_feature_control_info(control);
 	if (!ctl_info) {
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	if (mixer->protocol == UAC_VERSION_1)
@@ -1585,7 +1590,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
 
 	if (!kctl) {
 		usb_audio_err(mixer->chip, "cannot malloc kcontrol\n");
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	kctl->private_free = snd_usb_mixer_elem_free;
@@ -1755,7 +1760,7 @@ static void build_connector_control(struct usb_mixer_interface *mixer,
 	kctl = snd_ctl_new1(&usb_connector_ctl_ro, cval);
 	if (!kctl) {
 		usb_audio_err(mixer->chip, "cannot malloc kcontrol\n");
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	get_connector_control_name(mixer, term, is_input, kctl->id.name,
@@ -1808,7 +1813,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid,
 	kctl = snd_ctl_new1(&usb_bool_master_control_ctl_ro, cval);
 
 	if (!kctl) {
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return -ENOMEM;
 	}
 
@@ -2070,7 +2075,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state,
 	kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
 	if (!kctl) {
 		usb_audio_err(state->chip, "cannot malloc kcontrol\n");
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		return;
 	}
 	kctl->private_free = snd_usb_mixer_elem_free;
@@ -2468,7 +2473,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 
 		kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
 		if (!kctl) {
-			kfree(cval);
+			usb_mixer_elem_info_free(cval);
 			return -ENOMEM;
 		}
 		kctl->private_free = snd_usb_mixer_elem_free;
@@ -2606,7 +2611,7 @@ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl)
 	if (kctl->private_data) {
 		struct usb_mixer_elem_info *cval = kctl->private_data;
 		num_ins = cval->max;
-		kfree(cval);
+		usb_mixer_elem_info_free(cval);
 		kctl->private_data = NULL;
 	}
 	if (kctl->private_value) {
@@ -2678,10 +2683,10 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 		break;
 	}
 
-	namelist = kmalloc_array(desc->bNrInPins, sizeof(char *), GFP_KERNEL);
+	namelist = kcalloc(desc->bNrInPins, sizeof(char *), GFP_KERNEL);
 	if (!namelist) {
-		kfree(cval);
-		return -ENOMEM;
+		err = -ENOMEM;
+		goto error_cval;
 	}
 #define MAX_ITEM_NAME_LEN	64
 	for (i = 0; i < desc->bNrInPins; i++) {
@@ -2689,11 +2694,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 		len = 0;
 		namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL);
 		if (!namelist[i]) {
-			while (i--)
-				kfree(namelist[i]);
-			kfree(namelist);
-			kfree(cval);
-			return -ENOMEM;
+			err = -ENOMEM;
+			goto error_name;
 		}
 		len = check_mapped_selector_name(state, unitid, i, namelist[i],
 						 MAX_ITEM_NAME_LEN);
@@ -2707,10 +2709,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval);
 	if (! kctl) {
 		usb_audio_err(state->chip, "cannot malloc kcontrol\n");
-		for (i = 0; i < desc->bNrInPins; i++)
-			kfree(namelist[i]);
-		kfree(namelist);
-		kfree(cval);
+		err = -ENOMEM;
+		goto error_name;
 		return -ENOMEM;
 	}
 	kctl->private_value = (unsigned long)namelist;
@@ -2757,6 +2757,14 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	usb_audio_dbg(state->chip, "[%d] SU [%s] items = %d\n",
 		    cval->head.id, kctl->id.name, desc->bNrInPins);
 	return snd_usb_mixer_add_control(&cval->head, kctl);
+
+ error_name:
+	for (i = 0; i < desc->bNrInPins; i++)
+		kfree(namelist[i]);
+	kfree(namelist);
+ error_cval:
+	usb_mixer_elem_info_free(cval);
+	return err;
 }
 
 /*
-- 
cgit v1.2.3-59-g8ed1b


From b8e4f1fdfa422398c2d6c47bfb7d1feb3046d70a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 22 Aug 2019 09:25:27 +0200
Subject: ALSA: usb-audio: Remove superfluous bLength checks

Now that we got the more comprehensive validation code for USB-audio
descriptors, the check of overflow in each descriptor unit parser
became superfluous.  Drop some of the obvious cases.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/clock.c | 14 ++++------
 sound/usb/mixer.c | 84 -------------------------------------------------------
 2 files changed, 6 insertions(+), 92 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 72e9bdf76115..6b8c14f9b5d4 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -38,39 +38,37 @@ static void *find_uac_clock_desc(struct usb_host_interface *iface, int id,
 static bool validate_clock_source_v2(void *p, int id)
 {
 	struct uac_clock_source_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_source_v3(void *p, int id)
 {
 	struct uac3_clock_source_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_selector_v2(void *p, int id)
 {
 	struct uac_clock_selector_descriptor *cs = p;
-	return cs->bLength >= sizeof(*cs) && cs->bClockID == id &&
-		cs->bLength == 7 + cs->bNrInPins;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_selector_v3(void *p, int id)
 {
 	struct uac3_clock_selector_descriptor *cs = p;
-	return cs->bLength >= sizeof(*cs) && cs->bClockID == id &&
-		cs->bLength == 11 + cs->bNrInPins;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_multiplier_v2(void *p, int id)
 {
 	struct uac_clock_multiplier_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 static bool validate_clock_multiplier_v3(void *p, int id)
 {
 	struct uac3_clock_multiplier_descriptor *cs = p;
-	return cs->bLength == sizeof(*cs) && cs->bClockID == id;
+	return cs->bClockID == id;
 }
 
 #define DEFINE_FIND_HELPER(name, obj, validator, type)		\
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 277660fd6e0a..c9777d8a76f5 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -740,13 +740,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 {
 	int mu_channels;
 
-	if (desc->bLength < sizeof(*desc))
-		return -EINVAL;
-	if (!desc->bNrInPins)
-		return -EINVAL;
-	if (desc->bLength < sizeof(*desc) + desc->bNrInPins)
-		return -EINVAL;
-
 	switch (state->mixer->protocol) {
 	case UAC_VERSION_1:
 	case UAC_VERSION_2:
@@ -1781,13 +1774,6 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid,
 	if (state->mixer->protocol != UAC_VERSION_2)
 		return -EINVAL;
 
-	if (hdr->bLength != sizeof(*hdr)) {
-		usb_audio_dbg(state->chip,
-			      "Bogus clock source descriptor length of %d, ignoring.\n",
-			      hdr->bLength);
-		return 0;
-	}
-
 	/*
 	 * The only property of this unit we are interested in is the
 	 * clock source validity. If that isn't readable, just bail out.
@@ -1846,62 +1832,20 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid,
 	__u8 *bmaControls;
 
 	if (state->mixer->protocol == UAC_VERSION_1) {
-		if (hdr->bLength < 7) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 		csize = hdr->bControlSize;
-		if (!csize) {
-			usb_audio_dbg(state->chip,
-				      "unit %u: invalid bControlSize == 0\n",
-				      unitid);
-			return -EINVAL;
-		}
 		channels = (hdr->bLength - 7) / csize - 1;
 		bmaControls = hdr->bmaControls;
-		if (hdr->bLength < 7 + csize) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 	} else if (state->mixer->protocol == UAC_VERSION_2) {
 		struct uac2_feature_unit_descriptor *ftr = _ftr;
-		if (hdr->bLength < 6) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 		csize = 4;
 		channels = (hdr->bLength - 6) / 4 - 1;
 		bmaControls = ftr->bmaControls;
-		if (hdr->bLength < 6 + csize) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 	} else { /* UAC_VERSION_3 */
 		struct uac3_feature_unit_descriptor *ftr = _ftr;
 
-		if (hdr->bLength < 7) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 		csize = 4;
 		channels = (ftr->bLength - 7) / 4 - 1;
 		bmaControls = ftr->bmaControls;
-		if (hdr->bLength < 7 + csize) {
-			usb_audio_err(state->chip,
-				      "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n",
-				      unitid);
-			return -EINVAL;
-		}
 	}
 
 	/* parse the source unit */
@@ -2101,15 +2045,11 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid,
 
 	if (state->mixer->protocol == UAC_VERSION_2) {
 		struct uac2_input_terminal_descriptor *d_v2 = raw_desc;
-		if (d_v2->bLength < sizeof(*d_v2))
-			return -EINVAL;
 		control = UAC2_TE_CONNECTOR;
 		term_id = d_v2->bTerminalID;
 		bmctls = le16_to_cpu(d_v2->bmControls);
 	} else if (state->mixer->protocol == UAC_VERSION_3) {
 		struct uac3_input_terminal_descriptor *d_v3 = raw_desc;
-		if (d_v3->bLength < sizeof(*d_v3))
-			return -EINVAL;
 		control = UAC3_TE_INSERTION;
 		term_id = d_v3->bTerminalID;
 		bmctls = le32_to_cpu(d_v3->bmControls);
@@ -2371,18 +2311,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 	const char *name = extension_unit ?
 		"Extension Unit" : "Processing Unit";
 
-	if (desc->bLength < 13) {
-		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
-		return -EINVAL;
-	}
-
 	num_ins = desc->bNrInPins;
-	if (desc->bLength < 13 + num_ins ||
-	    desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) {
-		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
-		return -EINVAL;
-	}
-
 	for (i = 0; i < num_ins; i++) {
 		err = parse_audio_unit(state, desc->baSourceID[i]);
 		if (err < 0)
@@ -2637,13 +2566,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	const struct usbmix_name_map *map;
 	char **namelist;
 
-	if (desc->bLength < 5 || !desc->bNrInPins ||
-	    desc->bLength < 5 + desc->bNrInPins) {
-		usb_audio_err(state->chip,
-			"invalid SELECTOR UNIT descriptor %d\n", unitid);
-		return -EINVAL;
-	}
-
 	for (i = 0; i < desc->bNrInPins; i++) {
 		err = parse_audio_unit(state, desc->baSourceID[i]);
 		if (err < 0)
@@ -3149,8 +3071,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		if (mixer->protocol == UAC_VERSION_1) {
 			struct uac1_output_terminal_descriptor *desc = p;
 
-			if (desc->bLength < sizeof(*desc))
-				continue; /* invalid descriptor? */
 			/* mark terminal ID as visited */
 			set_bit(desc->bTerminalID, state.unitbitmap);
 			state.oterm.id = desc->bTerminalID;
@@ -3162,8 +3082,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		} else if (mixer->protocol == UAC_VERSION_2) {
 			struct uac2_output_terminal_descriptor *desc = p;
 
-			if (desc->bLength < sizeof(*desc))
-				continue; /* invalid descriptor? */
 			/* mark terminal ID as visited */
 			set_bit(desc->bTerminalID, state.unitbitmap);
 			state.oterm.id = desc->bTerminalID;
@@ -3189,8 +3107,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		} else {  /* UAC_VERSION_3 */
 			struct uac3_output_terminal_descriptor *desc = p;
 
-			if (desc->bLength < sizeof(*desc))
-				continue; /* invalid descriptor? */
 			/* mark terminal ID as visited */
 			set_bit(desc->bTerminalID, state.unitbitmap);
 			state.oterm.id = desc->bTerminalID;
-- 
cgit v1.2.3-59-g8ed1b


From 0ce6a624473e7e5752a84416f85f73d082308edd Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 22 Aug 2019 22:30:07 +0800
Subject: ASoC: AMD: Fix Kconfig warning without GPIOLIB

While do rand build without GPIOLIB, we get Kconfig warning:\

WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A
  Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && GPIOLIB [=n]
  Selected by [m]:
  - SND_SOC_AMD_CZ_DA7219MX98357_MACH [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP [=m] && I2C [=y]

Add GPIOLIB dependency to fix this.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190822143007.73644-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/Kconfig | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig
index 9ca9214cb7fb..5f40517717c4 100644
--- a/sound/soc/amd/Kconfig
+++ b/sound/soc/amd/Kconfig
@@ -10,7 +10,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH
 	select SND_SOC_MAX98357A
 	select SND_SOC_ADAU7002
 	select REGULATOR
-	depends on SND_SOC_AMD_ACP && I2C
+	depends on SND_SOC_AMD_ACP && I2C && GPIOLIB
 	help
 	 This option enables machine driver for DA7219 and MAX9835.
 
-- 
cgit v1.2.3-59-g8ed1b


From b9132b89933177286fb427bd03affcebc4d649e1 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 21 Aug 2019 11:47:29 -0500
Subject: ASoC: SOF: Add OF DSP device support

Add support for device tree based SOF DSP devices.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190821164730.7385-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/Kconfig      |  10 ++++
 sound/soc/sof/Makefile     |   3 +
 sound/soc/sof/sof-of-dev.c | 143 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 156 insertions(+)
 create mode 100644 sound/soc/sof/sof-of-dev.c

(limited to 'sound')

diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig
index fb01f0ca6027..01acb580b817 100644
--- a/sound/soc/sof/Kconfig
+++ b/sound/soc/sof/Kconfig
@@ -36,6 +36,16 @@ config SND_SOC_SOF_ACPI
 	  Say Y if you need this option
 	  If unsure select "N".
 
+config SND_SOC_SOF_OF
+	tristate "SOF OF enumeration support"
+	depends on OF || COMPILE_TEST
+	select SND_SOC_SOF
+	select SND_SOC_SOF_OPTIONS
+	help
+	  This adds support for Device Tree enumeration. This option is
+	  required to enable i.MX8 devices.
+	  Say Y if you need this option. If unsure select "N".
+
 config SND_SOC_SOF_OPTIONS
 	tristate
 	help
diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile
index 585fb6917489..772c452d1ae2 100644
--- a/sound/soc/sof/Makefile
+++ b/sound/soc/sof/Makefile
@@ -5,6 +5,8 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\
 
 snd-sof-pci-objs := sof-pci-dev.o
 snd-sof-acpi-objs := sof-acpi-dev.o
+snd-sof-of-objs := sof-of-dev.o
+
 snd-sof-nocodec-objs := nocodec.o
 
 obj-$(CONFIG_SND_SOC_SOF) += snd-sof.o
@@ -12,6 +14,7 @@ obj-$(CONFIG_SND_SOC_SOF_NOCODEC) += snd-sof-nocodec.o
 
 
 obj-$(CONFIG_SND_SOC_SOF_ACPI) += snd-sof-acpi.o
+obj-$(CONFIG_SND_SOC_SOF_OF) += snd-sof-of.o
 obj-$(CONFIG_SND_SOC_SOF_PCI) += snd-sof-pci.o
 
 obj-$(CONFIG_SND_SOC_SOF_INTEL_TOPLEVEL) += intel/
diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c
new file mode 100644
index 000000000000..28a9692974e5
--- /dev/null
+++ b/sound/soc/sof/sof-of-dev.c
@@ -0,0 +1,143 @@
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// Copyright 2019 NXP
+//
+// Author: Daniel Baluta <daniel.baluta@nxp.com>
+//
+
+#include <linux/firmware.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/sof.h>
+
+#include "ops.h"
+
+extern struct snd_sof_dsp_ops sof_imx8_ops;
+
+/* platform specific devices */
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8)
+static struct sof_dev_desc sof_of_imx8qxp_desc = {
+	.default_fw_path = "imx/sof",
+	.default_tplg_path = "imx/sof-tplg",
+	.nocodec_fw_filename = "sof-imx8.ri",
+	.nocodec_tplg_filename = "sof-imx8-nocodec.tplg",
+	.ops = &sof_imx8_ops,
+};
+#endif
+
+static const struct dev_pm_ops sof_of_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(snd_sof_suspend, snd_sof_resume)
+	SET_RUNTIME_PM_OPS(snd_sof_runtime_suspend, snd_sof_runtime_resume,
+			   NULL)
+};
+
+static void sof_of_probe_complete(struct device *dev)
+{
+	/* allow runtime_pm */
+	pm_runtime_set_autosuspend_delay(dev, SND_SOF_SUSPEND_DELAY_MS);
+	pm_runtime_use_autosuspend(dev);
+	pm_runtime_enable(dev);
+}
+
+static int sof_of_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	const struct sof_dev_desc *desc;
+	/*TODO: create a generic snd_soc_xxx_mach */
+	struct snd_soc_acpi_mach *mach;
+	struct snd_sof_pdata *sof_pdata;
+	const struct snd_sof_dsp_ops *ops;
+	int ret;
+
+	dev_info(&pdev->dev, "DT DSP detected");
+
+	sof_pdata = devm_kzalloc(dev, sizeof(*sof_pdata), GFP_KERNEL);
+	if (!sof_pdata)
+		return -ENOMEM;
+
+	desc = device_get_match_data(dev);
+	if (!desc)
+		return -ENODEV;
+
+	/* get ops for platform */
+	ops = desc->ops;
+	if (!ops) {
+		dev_err(dev, "error: no matching DT descriptor ops\n");
+		return -ENODEV;
+	}
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE)
+	/* force nocodec mode */
+	dev_warn(dev, "Force to use nocodec mode\n");
+	mach = devm_kzalloc(dev, sizeof(*mach), GFP_KERNEL);
+	if (!mach)
+		return -ENOMEM;
+	ret = sof_nocodec_setup(dev, sof_pdata, mach, desc, ops);
+	if (ret < 0)
+		return ret;
+#else
+	/* TODO: implement case where we actually have a codec */
+	return -ENODEV;
+#endif
+
+	if (mach)
+		mach->mach_params.platform = dev_name(dev);
+
+	sof_pdata->machine = mach;
+	sof_pdata->desc = desc;
+	sof_pdata->dev = &pdev->dev;
+	sof_pdata->platform = dev_name(dev);
+
+	/* TODO: read alternate fw and tplg filenames from DT */
+	sof_pdata->fw_filename_prefix = sof_pdata->desc->default_fw_path;
+	sof_pdata->tplg_filename_prefix = sof_pdata->desc->default_tplg_path;
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)
+	/* set callback to enable runtime_pm */
+	sof_pdata->sof_probe_complete = sof_of_probe_complete;
+#endif
+	 /* call sof helper for DSP hardware probe */
+	ret = snd_sof_device_probe(dev, sof_pdata);
+	if (ret) {
+		dev_err(dev, "error: failed to probe DSP hardware\n");
+		return ret;
+	}
+
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)
+	sof_of_probe_complete(dev);
+#endif
+
+	return ret;
+}
+
+static int sof_of_remove(struct platform_device *pdev)
+{
+	pm_runtime_disable(&pdev->dev);
+
+	/* call sof helper for DSP hardware remove */
+	snd_sof_device_remove(&pdev->dev);
+
+	return 0;
+}
+
+static const struct of_device_id sof_of_ids[] = {
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8)
+	{ .compatible = "fsl,imx8qxp-dsp", .data = &sof_of_imx8qxp_desc},
+#endif
+	{ }
+};
+MODULE_DEVICE_TABLE(of, sof_of_ids);
+
+/* DT driver definition */
+static struct platform_driver snd_sof_of_driver = {
+	.probe = sof_of_probe,
+	.remove = sof_of_remove,
+	.driver = {
+		.name = "sof-audio-of",
+		.pm = &sof_of_pm,
+		.of_match_table = sof_of_ids,
+	},
+};
+module_platform_driver(snd_sof_of_driver);
+
+MODULE_LICENSE("Dual BSD/GPL");
-- 
cgit v1.2.3-59-g8ed1b


From 202acc565a1f050f82fdab646fd9c86bfdb21733 Mon Sep 17 00:00:00 2001
From: Daniel Baluta <daniel.baluta@nxp.com>
Date: Wed, 21 Aug 2019 11:47:30 -0500
Subject: ASoC: SOF: imx: Add i.MX8 HW support

Add support for the audio DSP hardware found on NXP i.MX8 platform.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190821164730.7385-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/Kconfig      |   1 +
 sound/soc/sof/Makefile     |   1 +
 sound/soc/sof/imx/Kconfig  |  22 +++
 sound/soc/sof/imx/Makefile |   4 +
 sound/soc/sof/imx/imx8.c   | 394 +++++++++++++++++++++++++++++++++++++++++++++
 5 files changed, 422 insertions(+)
 create mode 100644 sound/soc/sof/imx/Kconfig
 create mode 100644 sound/soc/sof/imx/Makefile
 create mode 100644 sound/soc/sof/imx/imx8.c

(limited to 'sound')

diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig
index 01acb580b817..bb8036ae567e 100644
--- a/sound/soc/sof/Kconfig
+++ b/sound/soc/sof/Kconfig
@@ -173,6 +173,7 @@ config SND_SOC_SOF_PROBE_WORK_QUEUE
 	  When selected, the probe is handled in two steps, for example to
 	  avoid lockdeps if request_module is used in the probe.
 
+source "sound/soc/sof/imx/Kconfig"
 source "sound/soc/sof/intel/Kconfig"
 source "sound/soc/sof/xtensa/Kconfig"
 
diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile
index 772c452d1ae2..b0a6f01bdc44 100644
--- a/sound/soc/sof/Makefile
+++ b/sound/soc/sof/Makefile
@@ -18,4 +18,5 @@ obj-$(CONFIG_SND_SOC_SOF_OF) += snd-sof-of.o
 obj-$(CONFIG_SND_SOC_SOF_PCI) += snd-sof-pci.o
 
 obj-$(CONFIG_SND_SOC_SOF_INTEL_TOPLEVEL) += intel/
+obj-$(CONFIG_SND_SOC_SOF_IMX_TOPLEVEL) += imx/
 obj-$(CONFIG_SND_SOC_SOF_XTENSA) += xtensa/
diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig
new file mode 100644
index 000000000000..fd73d8402dbf
--- /dev/null
+++ b/sound/soc/sof/imx/Kconfig
@@ -0,0 +1,22 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+
+config SND_SOC_SOF_IMX_TOPLEVEL
+	bool "SOF support for NXP i.MX audio DSPs"
+	depends on ARM64 && SND_SOC_SOF_OF || COMPILE_TEST
+	help
+          This adds support for Sound Open Firmware for NXP i.MX platforms.
+          Say Y if you have such a device.
+          If unsure select "N".
+
+if SND_SOC_SOF_IMX_TOPLEVEL
+
+config SND_SOC_SOF_IMX8
+	tristate "SOF support for i.MX8"
+	depends on IMX_SCU
+	depends on IMX_DSP
+	help
+          This adds support for Sound Open Firmware for NXP i.MX8 platforms
+          Say Y if you have such a device.
+          If unsure select "N".
+
+endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL
diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile
new file mode 100644
index 000000000000..6ef908e8c807
--- /dev/null
+++ b/sound/soc/sof/imx/Makefile
@@ -0,0 +1,4 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+snd-sof-imx8-objs := imx8.o
+
+obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
new file mode 100644
index 000000000000..e502f584207f
--- /dev/null
+++ b/sound/soc/sof/imx/imx8.c
@@ -0,0 +1,394 @@
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// Copyright 2019 NXP
+//
+// Author: Daniel Baluta <daniel.baluta@nxp.com>
+//
+// Hardware interface for audio DSP on i.MX8
+
+#include <linux/firmware.h>
+#include <linux/of_platform.h>
+#include <linux/of_address.h>
+#include <linux/of_irq.h>
+#include <linux/pm_domain.h>
+
+#include <linux/module.h>
+#include <sound/sof.h>
+#include <sound/sof/xtensa.h>
+#include <linux/firmware/imx/ipc.h>
+#include <linux/firmware/imx/dsp.h>
+
+#include <linux/firmware/imx/svc/misc.h>
+#include <dt-bindings/firmware/imx/rsrc.h>
+#include "../ops.h"
+
+/* DSP memories */
+#define IRAM_OFFSET		0x10000
+#define IRAM_SIZE		(2 * 1024)
+#define DRAM0_OFFSET		0x0
+#define DRAM0_SIZE		(32 * 1024)
+#define DRAM1_OFFSET		0x8000
+#define DRAM1_SIZE		(32 * 1024)
+#define SYSRAM_OFFSET		0x18000
+#define SYSRAM_SIZE		(256 * 1024)
+#define SYSROM_OFFSET		0x58000
+#define SYSROM_SIZE		(192 * 1024)
+
+#define RESET_VECTOR_VADDR	0x596f8000
+
+#define MBOX_OFFSET	0x800000
+#define MBOX_SIZE	0x1000
+
+struct imx8_priv {
+	struct device *dev;
+	struct snd_sof_dev *sdev;
+
+	/* DSP IPC handler */
+	struct imx_dsp_ipc *dsp_ipc;
+	struct platform_device *ipc_dev;
+
+	/* System Controller IPC handler */
+	struct imx_sc_ipc *sc_ipc;
+
+	/* Power domain handling */
+	int num_domains;
+	struct device **pd_dev;
+	struct device_link **link;
+
+};
+
+static void imx8_get_reply(struct snd_sof_dev *sdev)
+{
+	struct snd_sof_ipc_msg *msg = sdev->msg;
+	struct sof_ipc_reply reply;
+	int ret = 0;
+
+	if (!msg) {
+		dev_warn(sdev->dev, "unexpected ipc interrupt\n");
+		return;
+	}
+
+	/* get reply */
+	sof_mailbox_read(sdev, sdev->host_box.offset, &reply, sizeof(reply));
+
+	if (reply.error < 0) {
+		memcpy(msg->reply_data, &reply, sizeof(reply));
+		ret = reply.error;
+	} else {
+		/* reply has correct size? */
+		if (reply.hdr.size != msg->reply_size) {
+			dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n",
+				msg->reply_size, reply.hdr.size);
+			ret = -EINVAL;
+		}
+
+		/* read the message */
+		if (msg->reply_size > 0)
+			sof_mailbox_read(sdev, sdev->host_box.offset,
+					 msg->reply_data, msg->reply_size);
+	}
+
+	msg->reply_error = ret;
+}
+
+static int imx8_get_mailbox_offset(struct snd_sof_dev *sdev)
+{
+	return MBOX_OFFSET;
+}
+
+static int imx8_get_window_offset(struct snd_sof_dev *sdev, u32 id)
+{
+	return MBOX_OFFSET;
+}
+
+void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc)
+{
+	struct imx8_priv *priv = imx_dsp_get_data(ipc);
+	unsigned long flags;
+
+	spin_lock_irqsave(&priv->sdev->ipc_lock, flags);
+	imx8_get_reply(priv->sdev);
+	snd_sof_ipc_reply(priv->sdev, 0);
+	spin_unlock_irqrestore(&priv->sdev->ipc_lock, flags);
+}
+
+void imx8_dsp_handle_request(struct imx_dsp_ipc *ipc)
+{
+	struct imx8_priv *priv = imx_dsp_get_data(ipc);
+
+	snd_sof_ipc_msgs_rx(priv->sdev);
+}
+
+struct imx_dsp_ops dsp_ops = {
+	.handle_reply		= imx8_dsp_handle_reply,
+	.handle_request		= imx8_dsp_handle_request,
+};
+
+static int imx8_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
+{
+	struct imx8_priv *priv = (struct imx8_priv *)sdev->private;
+
+	sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data,
+			  msg->msg_size);
+	imx_dsp_ring_doorbell(priv->dsp_ipc, 0);
+
+	return 0;
+}
+
+/*
+ * DSP control.
+ */
+static int imx8_run(struct snd_sof_dev *sdev)
+{
+	struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private;
+	int ret;
+
+	ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP,
+				      IMX_SC_C_OFS_SEL, 1);
+	if (ret < 0) {
+		dev_err(sdev->dev, "Error system address offset source select\n");
+		return ret;
+	}
+
+	ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP,
+				      IMX_SC_C_OFS_AUDIO, 0x80);
+	if (ret < 0) {
+		dev_err(sdev->dev, "Error system address offset of AUDIO\n");
+		return ret;
+	}
+
+	ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP,
+				      IMX_SC_C_OFS_PERIPH, 0x5A);
+	if (ret < 0) {
+		dev_err(sdev->dev, "Error system address offset of PERIPH %d\n",
+			ret);
+		return ret;
+	}
+
+	ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP,
+				      IMX_SC_C_OFS_IRQ, 0x51);
+	if (ret < 0) {
+		dev_err(sdev->dev, "Error system address offset of IRQ\n");
+		return ret;
+	}
+
+	imx_sc_pm_cpu_start(dsp_priv->sc_ipc, IMX_SC_R_DSP, true,
+			    RESET_VECTOR_VADDR);
+
+	return 0;
+}
+
+static int imx8_probe(struct snd_sof_dev *sdev)
+{
+	struct platform_device *pdev =
+		container_of(sdev->dev, struct platform_device, dev);
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *res_node;
+	struct resource *mmio;
+	struct imx8_priv *priv;
+	struct resource res;
+	u32 base, size;
+	int ret = 0;
+	int i;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	sdev->private = priv;
+	priv->dev = sdev->dev;
+	priv->sdev = sdev;
+
+	/* power up device associated power domains */
+	priv->num_domains = of_count_phandle_with_args(np, "power-domains",
+						       "#power-domain-cells");
+	if (priv->num_domains < 0) {
+		dev_err(sdev->dev, "no power-domains property in %pOF\n", np);
+		return priv->num_domains;
+	}
+
+	priv->pd_dev = devm_kmalloc_array(&pdev->dev, priv->num_domains,
+					  sizeof(*priv->pd_dev), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->link = devm_kmalloc_array(&pdev->dev, priv->num_domains,
+					sizeof(*priv->link), GFP_KERNEL);
+	if (!priv->link)
+		return -ENOMEM;
+
+	for (i = 0; i < priv->num_domains; i++) {
+		priv->pd_dev[i] = dev_pm_domain_attach_by_id(&pdev->dev, i);
+		if (IS_ERR(priv->pd_dev[i])) {
+			ret = PTR_ERR(priv->pd_dev[i]);
+			goto exit_unroll_pm;
+		}
+		priv->link[i] = device_link_add(&pdev->dev, priv->pd_dev[i],
+						DL_FLAG_STATELESS |
+						DL_FLAG_PM_RUNTIME |
+						DL_FLAG_RPM_ACTIVE);
+		if (IS_ERR(priv->link[i])) {
+			ret = PTR_ERR(priv->link[i]);
+			dev_pm_domain_detach(priv->pd_dev[i], false);
+			goto exit_unroll_pm;
+		}
+	}
+
+	ret = imx_scu_get_handle(&priv->sc_ipc);
+	if (ret) {
+		dev_err(sdev->dev, "Cannot obtain SCU handle (err = %d)\n",
+			ret);
+		goto exit_unroll_pm;
+	}
+
+	priv->ipc_dev = platform_device_register_data(sdev->dev, "imx-dsp",
+						      PLATFORM_DEVID_NONE,
+						      pdev, sizeof(*pdev));
+	if (IS_ERR(priv->ipc_dev)) {
+		ret = PTR_ERR(priv->ipc_dev);
+		goto exit_unroll_pm;
+	}
+
+	priv->dsp_ipc = dev_get_drvdata(&priv->ipc_dev->dev);
+	if (!priv->dsp_ipc) {
+		/* DSP IPC driver not probed yet, try later */
+		ret = -EPROBE_DEFER;
+		dev_err(sdev->dev, "Failed to get drvdata\n");
+		goto exit_pdev_unregister;
+	}
+
+	imx_dsp_set_data(priv->dsp_ipc, priv);
+	priv->dsp_ipc->ops = &dsp_ops;
+
+	/* DSP base */
+	mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (mmio) {
+		base = mmio->start;
+		size = resource_size(mmio);
+	} else {
+		dev_err(sdev->dev, "error: failed to get DSP base at idx 0\n");
+		ret = -EINVAL;
+		goto exit_pdev_unregister;
+	}
+
+	sdev->bar[SOF_FW_BLK_TYPE_IRAM] = devm_ioremap(sdev->dev, base, size);
+	if (!sdev->bar[SOF_FW_BLK_TYPE_IRAM]) {
+		dev_err(sdev->dev, "failed to ioremap base 0x%x size 0x%x\n",
+			base, size);
+		ret = -ENODEV;
+		goto exit_pdev_unregister;
+	}
+	sdev->mmio_bar = SOF_FW_BLK_TYPE_IRAM;
+
+	res_node = of_parse_phandle(np, "memory-region", 0);
+	if (!res_node) {
+		dev_err(&pdev->dev, "failed to get memory region node\n");
+		ret = -ENODEV;
+		goto exit_pdev_unregister;
+	}
+
+	ret = of_address_to_resource(res_node, 0, &res);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to get reserved region address\n");
+		goto exit_pdev_unregister;
+	}
+
+	sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, res.start,
+							  res.end - res.start +
+							  1);
+	if (IS_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM])) {
+		dev_err(sdev->dev, "failed to ioremap mem 0x%x size 0x%x\n",
+			base, size);
+		ret = PTR_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM]);
+		goto exit_pdev_unregister;
+	}
+	sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM;
+
+	return 0;
+
+exit_pdev_unregister:
+	platform_device_unregister(priv->ipc_dev);
+exit_unroll_pm:
+	while (--i >= 0) {
+		device_link_del(priv->link[i]);
+		dev_pm_domain_detach(priv->pd_dev[i], false);
+	}
+
+	return ret;
+}
+
+static int imx8_remove(struct snd_sof_dev *sdev)
+{
+	struct imx8_priv *priv = (struct imx8_priv *)sdev->private;
+	int i;
+
+	platform_device_unregister(priv->ipc_dev);
+
+	for (i = 0; i < priv->num_domains; i++) {
+		device_link_del(priv->link[i]);
+		dev_pm_domain_detach(priv->pd_dev[i], false);
+	}
+
+	return 0;
+}
+
+/* on i.MX8 there is 1 to 1 match between type and BAR idx */
+int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type)
+{
+	return type;
+}
+
+void imx8_ipc_msg_data(struct snd_sof_dev *sdev,
+		       struct snd_pcm_substream *substream,
+		       void *p, size_t sz)
+{
+	sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz);
+}
+
+int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
+			struct snd_pcm_substream *substream,
+			const struct sof_ipc_pcm_params_reply *reply)
+{
+	return 0;
+}
+
+static struct snd_soc_dai_driver imx8_dai[] = {
+{
+	.name = "esai-port",
+},
+};
+
+/* i.MX8  ops */
+struct snd_sof_dsp_ops sof_imx8_ops = {
+	/* probe and remove */
+	.probe		= imx8_probe,
+	.remove		= imx8_remove,
+	/* DSP core boot */
+	.run		= imx8_run,
+
+	/* Block IO */
+	.block_read	= sof_block_read,
+	.block_write	= sof_block_write,
+
+	/* ipc */
+	.send_msg	= imx8_send_msg,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset	= imx8_get_mailbox_offset,
+	.get_window_offset	= imx8_get_window_offset,
+
+	.ipc_msg_data	= imx8_ipc_msg_data,
+	.ipc_pcm_params	= imx8_ipc_pcm_params,
+
+	/* module loading */
+	.load_module	= snd_sof_parse_module_memcpy,
+	.get_bar_index	= imx8_get_bar_index,
+	/* firmware loading */
+	.load_firmware	= snd_sof_load_firmware_memcpy,
+
+	/* DAI drivers */
+	.drv = imx8_dai,
+	.num_drv = 1, /* we have only 1 ESAI interface on i.MX8 */
+};
+EXPORT_SYMBOL(sof_imx8_ops);
+
+MODULE_LICENSE("Dual BSD/GPL");
-- 
cgit v1.2.3-59-g8ed1b


From c101fb29737f4558bf589d0d66371d9e21040568 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 22 Aug 2019 22:36:08 +0800
Subject: ASoC: wm8988: fix typo in wm8988_right_line_controls

sound/soc/codecs/wm8988.c:270:30: warning:
 wm8988_rline_enum defined but not used [-Wunused-const-variable=]

wm8988_rline_enum should be used in wm8988_right_line_controls.

Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 5409fb4e327a ("ASoC: Add WM8988 CODEC driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190822143608.59824-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8988.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 25e74cf0666a..85bfd041d546 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -273,7 +273,7 @@ static const struct soc_enum wm8988_rline_enum =
 			      wm8988_line_texts,
 			      wm8988_line_values);
 static const struct snd_kcontrol_new wm8988_right_line_controls =
-	SOC_DAPM_ENUM("Route", wm8988_lline_enum);
+	SOC_DAPM_ENUM("Route", wm8988_rline_enum);
 
 /* Left Mixer */
 static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = {
-- 
cgit v1.2.3-59-g8ed1b


From 97aad5ce88164bc5f884b5234f015ac553ea2fe5 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 22 Aug 2019 22:37:47 +0800
Subject: ASoC: mediatek: mt2701: Fix -Wunused-const-variable warnings

sound/soc/mediatek/mt2701/mt2701-afe-common.h:66:27: warning:
 mt2701_afe_backup_list defined but not used [-Wunused-const-variable=]

mt2701_afe_backup_list is only used in mt2701-afe-pcm.c,
so just move the definition over there.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190822143747.20944-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt2701/mt2701-afe-common.h | 21 ---------------------
 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c    | 21 +++++++++++++++++++++
 2 files changed, 21 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-common.h b/sound/soc/mediatek/mt2701/mt2701-afe-common.h
index d44faba27d3c..32bef5e2a56d 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-common.h
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-common.h
@@ -63,27 +63,6 @@ enum audio_base_clock {
 	MT2701_BASE_CLK_NUM,
 };
 
-static const unsigned int mt2701_afe_backup_list[] = {
-	AUDIO_TOP_CON0,
-	AUDIO_TOP_CON4,
-	AUDIO_TOP_CON5,
-	ASYS_TOP_CON,
-	AFE_CONN0,
-	AFE_CONN1,
-	AFE_CONN2,
-	AFE_CONN3,
-	AFE_CONN15,
-	AFE_CONN16,
-	AFE_CONN17,
-	AFE_CONN18,
-	AFE_CONN19,
-	AFE_CONN20,
-	AFE_CONN21,
-	AFE_CONN22,
-	AFE_DAC_CON0,
-	AFE_MEMIF_PBUF_SIZE,
-};
-
 struct mt2701_i2s_data {
 	int i2s_ctrl_reg;
 	int i2s_asrc_fs_shift;
diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
index d7f5defa50c2..76502ba261c8 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
@@ -60,6 +60,27 @@ static const struct mt2701_afe_rate mt2701_afe_i2s_rates[] = {
 	{ .rate = 352800, .regvalue = 24 },
 };
 
+static const unsigned int mt2701_afe_backup_list[] = {
+	AUDIO_TOP_CON0,
+	AUDIO_TOP_CON4,
+	AUDIO_TOP_CON5,
+	ASYS_TOP_CON,
+	AFE_CONN0,
+	AFE_CONN1,
+	AFE_CONN2,
+	AFE_CONN3,
+	AFE_CONN15,
+	AFE_CONN16,
+	AFE_CONN17,
+	AFE_CONN18,
+	AFE_CONN19,
+	AFE_CONN20,
+	AFE_CONN21,
+	AFE_CONN22,
+	AFE_DAC_CON0,
+	AFE_MEMIF_PBUF_SIZE,
+};
+
 static int mt2701_dai_num_to_i2s(struct mtk_base_afe *afe, int num)
 {
 	struct mt2701_afe_private *afe_priv = afe->platform_priv;
-- 
cgit v1.2.3-59-g8ed1b


From 6191cbde5ff02ef5307e7b2e4f5b8f7607c2b1df Mon Sep 17 00:00:00 2001
From: Jiaxin Yu <jiaxin.yu@mediatek.com>
Date: Thu, 22 Aug 2019 20:51:00 +0800
Subject: ASoC: mediatek: mt8183: switch tdm pins gpio function when playback
 on or off

Pull TDM pins down when TDM BE shutdown to avoid current leakage.

Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1566478261-13464-2-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../mt8183/mt8183-mt6358-ts3a227-max98357.c        | 115 +++++++++++++++++++--
 1 file changed, 109 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 272766c1b859..fe8a7461ace5 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -46,6 +46,32 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
+static const struct snd_soc_dapm_widget
+mt8183_mt6358_ts3a227_max98357_dapm_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("IT6505_8CH"),
+};
+
+static const struct snd_soc_dapm_route
+mt8183_mt6358_ts3a227_max98357_dapm_routes[] = {
+	{"IT6505_8CH", NULL, "TDM"},
+};
+
+enum PINCTRL_PIN_STATE {
+	PIN_STATE_DEFAULT = 0,
+	PIN_TDM_OUT_ON,
+	PIN_TDM_OUT_OFF,
+	PIN_STATE_MAX
+};
+
+static const char * const mt8183_pin_str[PIN_STATE_MAX] = {
+	"default", "aud_tdm_out_on", "aud_tdm_out_off",
+};
+
+struct mt8183_mt6358_ts3a227_max98357_priv {
+	struct pinctrl *pinctrl;
+	struct pinctrl_state *pin_states[PIN_STATE_MAX];
+};
+
 static int
 mt8183_mt6358_ts3a227_max98357_bt_sco_startup(
 	struct snd_pcm_substream *substream)
@@ -173,6 +199,47 @@ SND_SOC_DAILINK_DEFS(tdm,
 	DAILINK_COMP_ARRAY(COMP_DUMMY()),
 	DAILINK_COMP_ARRAY(COMP_EMPTY()));
 
+static int mt8183_mt6358_tdm_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct mt8183_mt6358_ts3a227_max98357_priv *priv =
+		snd_soc_card_get_drvdata(rtd->card);
+	int ret;
+
+	if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON]))
+		return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]);
+
+	ret = pinctrl_select_state(priv->pinctrl,
+				   priv->pin_states[PIN_TDM_OUT_ON]);
+	if (ret)
+		dev_err(rtd->card->dev, "%s failed to select state %d\n",
+			__func__, ret);
+
+	return ret;
+}
+
+static void mt8183_mt6358_tdm_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct mt8183_mt6358_ts3a227_max98357_priv *priv =
+		snd_soc_card_get_drvdata(rtd->card);
+	int ret;
+
+	if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF]))
+		return;
+
+	ret = pinctrl_select_state(priv->pinctrl,
+				   priv->pin_states[PIN_TDM_OUT_OFF]);
+	if (ret)
+		dev_err(rtd->card->dev, "%s failed to select state %d\n",
+			__func__, ret);
+}
+
+static struct snd_soc_ops mt8183_mt6358_tdm_ops = {
+	.startup = mt8183_mt6358_tdm_startup,
+	.shutdown = mt8183_mt6358_tdm_shutdown,
+};
+
 static struct snd_soc_dai_link
 mt8183_mt6358_ts3a227_max98357_dai_links[] = {
 	/* FE */
@@ -325,6 +392,8 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = {
 		.no_pcm = 1,
 		.dpcm_playback = 1,
 		.ignore_suspend = 1,
+		.be_hw_params_fixup = mt8183_i2s_hw_params_fixup,
+		.ops = &mt8183_mt6358_tdm_ops,
 		SND_SOC_DAILINK_REG(tdm),
 	},
 };
@@ -371,7 +440,8 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev)
 	struct snd_soc_card *card = &mt8183_mt6358_ts3a227_max98357_card;
 	struct device_node *platform_node;
 	struct snd_soc_dai_link *dai_link;
-	struct pinctrl *default_pins;
+	struct mt8183_mt6358_ts3a227_max98357_priv *priv;
+	int ret;
 	int i;
 
 	card->dev = &pdev->dev;
@@ -397,12 +467,45 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev)
 		card->num_aux_devs = 1;
 	}
 
-	default_pins =
-		devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT);
-	if (IS_ERR(default_pins)) {
-		dev_err(&pdev->dev, "%s set pins failed\n",
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	snd_soc_card_set_drvdata(card, priv);
+
+	priv->pinctrl = devm_pinctrl_get(&pdev->dev);
+	if (IS_ERR(priv->pinctrl)) {
+		dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n",
 			__func__);
-		return PTR_ERR(default_pins);
+		return PTR_ERR(priv->pinctrl);
+	}
+
+	for (i = 0 ; i < PIN_STATE_MAX ; i++) {
+		priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl,
+							   mt8183_pin_str[i]);
+		if (IS_ERR(priv->pin_states[i])) {
+			ret = PTR_ERR(priv->pin_states[i]);
+			dev_info(&pdev->dev, "%s Can't find pin state %s %d\n",
+				 __func__, mt8183_pin_str[i], ret);
+		}
+	}
+
+	if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) {
+		ret = pinctrl_select_state(priv->pinctrl,
+					   priv->pin_states[PIN_TDM_OUT_OFF]);
+		if (ret)
+			dev_info(&pdev->dev,
+				 "%s failed to select state %d\n",
+				 __func__, ret);
+	}
+
+	if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) {
+		ret = pinctrl_select_state(priv->pinctrl,
+					   priv->pin_states[PIN_STATE_DEFAULT]);
+		if (ret)
+			dev_info(&pdev->dev,
+				 "%s failed to select state %d\n",
+				 __func__, ret);
 	}
 
 	return devm_snd_soc_register_card(&pdev->dev, card);
-- 
cgit v1.2.3-59-g8ed1b


From 4168ddabb480bef818c93f378428632fb681b500 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:04:49 +0900
Subject: ASoC: soc-core: use device_register()

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.

soc-core.c is using device_unregiser(), but there is no its paired
device_regiser(). We can find its code at soc_post_component_init()
which is using device_initialize() and device_add().
Here, device_initialize() + device_add() = device_register().

	-- linux/drivers/base/core.c --
	int device_register(struct device *dev)
	{
		device_initialize(dev);
		return device_add(dev);
	}

device_initialize() is doing each dev member's initialization only,
not related to device parent/release/groups.
Thus, we can postpone it.
let's use device_register() instead of device_initialize()/device_add().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/878sro1ldw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6df880be1622..3860d8521734 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1347,7 +1347,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 	rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL);
 	if (!rtd->dev)
 		return -ENOMEM;
-	device_initialize(rtd->dev);
 	rtd->dev->parent = rtd->card->dev;
 	rtd->dev->release = rtd_release;
 	rtd->dev->groups = soc_dev_attr_groups;
@@ -1357,7 +1356,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
-	ret = device_add(rtd->dev);
+	ret = device_register(rtd->dev);
 	if (ret < 0) {
 		/* calling put_device() here to free the rtd->dev */
 		put_device(rtd->dev);
-- 
cgit v1.2.3-59-g8ed1b


From a33c0d166cc5bcb3b9718649b84974216709acb1 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:05:02 +0900
Subject: ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup()

We can find specified name component via snd_soc_rtdcom_lookup().
But, it is not enough under multi CPU/Codec/Platform, because many
components which have same driver name might be connected to same rtd.

Not using this function as much as possible is best solution,
but some drivers are already deeply depended to it.

We can expand this function, for example having "num" which specifies
found order at parameter, etc (In such case, it need to have fixed
probing order).
Or, use different driver name in such component, etc.

We will have such issue if multi CPU/Codec/Platform were supported.
To indicate it, this patch adds NOTE to this function.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/874l2c1ldi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3860d8521734..0ed6576bfef4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -315,6 +315,14 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd,
 	if (!driver_name)
 		return NULL;
 
+	/*
+	 * NOTE
+	 *
+	 * snd_soc_rtdcom_lookup() will find component from rtd by using
+	 * specified driver name.
+	 * But, if many components which have same driver name are connected
+	 * to 1 rtd, this function will return 1st found component.
+	 */
 	for_each_rtdcom(rtd, rtdcom) {
 		const char *component_name = rtdcom->component->driver->name;
 
-- 
cgit v1.2.3-59-g8ed1b


From a4de83a385670c22c31e9bbb726595a447b32ba4 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:05:10 +0900
Subject: ASoC: soc-core: soc_cleanup_card_resources() become void

There is no need to check return value for
soc_cleanup_card_resources(). Let't makes it as void.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/871rxg1lda.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0ed6576bfef4..1be069c2ac8d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1885,7 +1885,7 @@ match:
 	}
 }
 
-static int soc_cleanup_card_resources(struct snd_soc_card *card)
+static void soc_cleanup_card_resources(struct snd_soc_card *card)
 {
 	/* free the ALSA card at first; this syncs with pending operations */
 	if (card->snd_card) {
@@ -1906,8 +1906,6 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card)
 	/* remove the card */
 	if (card->remove)
 		card->remove(card);
-
-	return 0;
 }
 
 static int snd_soc_instantiate_card(struct snd_soc_card *card)
-- 
cgit v1.2.3-59-g8ed1b


From 43ca5dab978294eae26a36f8989b6f0769da4256 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:05:32 +0900
Subject: ASoC: soc-topology: use for_each_component_dais() at remove_dai()

commit 52abe6cc1866a ("ASoC: topology: fix oops/use-after-free case
with dai driver") fixups remove_dai() error, but it is using
list_for_each_entry() for component->dai_list.

We already have for_each_component_dais() macro for it.
Let's use exising method.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tvaczazd.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-topology.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index dc463f1a9e24..b8690715abb5 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -530,7 +530,7 @@ static void remove_dai(struct snd_soc_component *comp,
 	if (dobj->ops && dobj->ops->dai_unload)
 		dobj->ops->dai_unload(comp, dobj);
 
-	list_for_each_entry(dai, &comp->dai_list, list)
+	for_each_component_dais(comp, dai)
 		if (dai->driver == dai_drv)
 			dai->driver = NULL;
 
-- 
cgit v1.2.3-59-g8ed1b


From 8d92bb516831e80fac916701447ee6e9f0a6f0f2 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:05:16 +0900
Subject: ASoC: soc-core: initialize component list

It might return without initializing in error case.
In such case, uninitialized variable might be used at error handler.
This patch initializes all necessary variable before return.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87zhk4zazt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1be069c2ac8d..1a17cb1bc03b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2647,6 +2647,9 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 {
 	struct snd_soc_dapm_context *dapm;
 
+	INIT_LIST_HEAD(&component->dai_list);
+	mutex_init(&component->io_mutex);
+
 	component->name = fmt_single_name(dev, &component->id);
 	if (!component->name) {
 		dev_err(dev, "ASoC: Failed to allocate name\n");
@@ -2663,9 +2666,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	dapm->idle_bias_off = !driver->idle_bias_on;
 	dapm->suspend_bias_off = driver->suspend_bias_off;
 
-	INIT_LIST_HEAD(&component->dai_list);
-	mutex_init(&component->io_mutex);
-
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From b2f3e0c9e4ee2e07fd880cc0c4da20c42d633d4b Mon Sep 17 00:00:00 2001
From: Jaska Uimonen <jaska.uimonen@intel.com>
Date: Thu, 22 Aug 2019 00:11:38 +0300
Subject: ASoC: SOF: topology: fix get control data return type and arguments

sof_get_control_data returns negative values even though the return
value is defined unsigned (size_t). So change the return value type to
int and add the data size as pointer argument to sof_get_control_data to
avoid ambiquity in the meaning of the return type.

Fixes: cac974a51ebb ("ASoC: SOF: topology: use set_get_data in process load")
Reported by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Jaska Uimonen <jaska.uimonen@intel.com>
Link: https://lore.kernel.org/r/20190821211138.14618-1-jaska.uimonen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/topology.c | 21 +++++++++++----------
 1 file changed, 11 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 28a7a6e06a53..fc85efbad378 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -1752,17 +1752,19 @@ err:
 	return ret;
 }
 
-static size_t sof_get_control_data(struct snd_sof_dev *sdev,
-				   struct snd_soc_dapm_widget *widget,
-				   struct sof_widget_data *wdata)
+static int sof_get_control_data(struct snd_sof_dev *sdev,
+				struct snd_soc_dapm_widget *widget,
+				struct sof_widget_data *wdata,
+				size_t *size)
 {
 	const struct snd_kcontrol_new *kc;
 	struct soc_mixer_control *sm;
 	struct soc_bytes_ext *sbe;
 	struct soc_enum *se;
-	size_t size = 0;
 	int i;
 
+	*size = 0;
+
 	for (i = 0; i < widget->num_kcontrols; i++) {
 		kc = &widget->kcontrol_news[i];
 
@@ -1800,7 +1802,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev,
 		if (wdata[i].pdata->magic != SOF_ABI_MAGIC)
 			return -EINVAL;
 
-		size += wdata[i].pdata->size;
+		*size += wdata[i].pdata->size;
 
 		/* get data type */
 		switch (wdata[i].control->cmd) {
@@ -1819,7 +1821,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev,
 		}
 	}
 
-	return size;
+	return 0;
 }
 
 static int sof_process_load(struct snd_soc_component *scomp, int index,
@@ -1855,12 +1857,11 @@ static int sof_process_load(struct snd_soc_component *scomp, int index,
 			return -ENOMEM;
 
 		/* get possible component controls and get size of all pdata */
-		ipc_data_size = sof_get_control_data(sdev, widget, wdata);
+		ret = sof_get_control_data(sdev, widget, wdata,
+					   &ipc_data_size);
 
-		if (ipc_data_size <= 0) {
-			ret = ipc_data_size;
+		if (ret < 0)
 			goto out;
-		}
 	}
 
 	ipc_size = sizeof(struct sof_ipc_comp_process) +
-- 
cgit v1.2.3-59-g8ed1b


From 495efdb01f89a5fc53f9b2e61f5726d804d4a15d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 20 Aug 2019 14:05:20 +0900
Subject: ASoC: soc-core: initialize list at one place

Initialize component related list at random place is very difficult
to read. This patch initialize it at snd_soc_component_initialize().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87y2zozazp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1a17cb1bc03b..0af83963289f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1272,7 +1272,6 @@ static int soc_probe_component(struct snd_soc_card *card,
 
 	component->card = card;
 	dapm->card = card;
-	INIT_LIST_HEAD(&component->card_list);
 	INIT_LIST_HEAD(&dapm->list);
 	soc_set_name_prefix(card, component);
 
@@ -2648,6 +2647,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	struct snd_soc_dapm_context *dapm;
 
 	INIT_LIST_HEAD(&component->dai_list);
+	INIT_LIST_HEAD(&component->dobj_list);
+	INIT_LIST_HEAD(&component->card_list);
 	mutex_init(&component->io_mutex);
 
 	component->name = fmt_single_name(dev, &component->id);
@@ -2733,7 +2734,6 @@ static void snd_soc_component_add(struct snd_soc_component *component)
 
 	/* see for_each_component */
 	list_add(&component->list, &component_list);
-	INIT_LIST_HEAD(&component->dobj_list);
 
 	mutex_unlock(&client_mutex);
 }
-- 
cgit v1.2.3-59-g8ed1b


From 542694df7511977c3baa6ba855126a0cce883977 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 23 Aug 2019 09:58:32 +0900
Subject: ASoC: soc-core: rename soc_post_component_init() to soc_rtd_init()

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

From function name point of view, "soc_post_component_init()" sounds
like "component initialize function".
But in reality it is rtd setup function.

This patch renames soc_post_component_init() to soc_rtd_init()

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v9uo7lc3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 9 ++++-----
 1 file changed, 4 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0af83963289f..3c087b478398 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1340,13 +1340,12 @@ err_probe:
 	return ret;
 }
 
-static void rtd_release(struct device *dev)
+static void soc_rtd_release(struct device *dev)
 {
 	kfree(dev);
 }
 
-static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
-	const char *name)
+static int soc_rtd_init(struct snd_soc_pcm_runtime *rtd, const char *name)
 {
 	int ret = 0;
 
@@ -1355,7 +1354,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 	if (!rtd->dev)
 		return -ENOMEM;
 	rtd->dev->parent = rtd->card->dev;
-	rtd->dev->release = rtd_release;
+	rtd->dev->release = soc_rtd_release;
 	rtd->dev->groups = soc_dev_attr_groups;
 	dev_set_name(rtd->dev, "%s", name);
 	dev_set_drvdata(rtd->dev, rtd);
@@ -1483,7 +1482,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 			return ret;
 	}
 
-	ret = soc_post_component_init(rtd, dai_link->name);
+	ret = soc_rtd_init(rtd, dai_link->name);
 	if (ret)
 		return ret;
 
-- 
cgit v1.2.3-59-g8ed1b


From e0ccdef92653f8867e2d1667facfd3c23699f540 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 23 Aug 2019 12:38:07 +0200
Subject: ALSA: usb-audio: Clean up check_input_term()

The primary changes in this patch are cleanups of __check_input_term()
and move to a non-nested switch-case block by evaluating the pair of
UAC version and the unit type, as we've done for parse_audio_unit().
Also each parser is split into the function for readability.

Now, a slight behavior change by this cleanup is the handling of
processing and extension units.  Formerly we've dealt with them
differently between UAC1/2 and UAC3; the latter returns an error if no
input sources are available, while the former continues to parse.

In this patch, unify the behavior in all cases: when input sources are
available, it parses recursively, then override the type and the id,
as well as channel information if not provided yet.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 401 ++++++++++++++++++++++++++++--------------------------
 1 file changed, 209 insertions(+), 192 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index a0468c4a6326..83d22f241bdd 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -758,224 +758,242 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 }
 
 /*
- * parse the source unit recursively until it reaches to a terminal
- * or a branched unit.
+ * Parse Input Terminal Unit
  */
 static int __check_input_term(struct mixer_build *state, int id,
-			    struct usb_audio_term *term)
+			      struct usb_audio_term *term);
+
+static int parse_term_uac1_iterm_unit(struct mixer_build *state,
+				      struct usb_audio_term *term,
+				      void *p1, int id)
 {
-	int protocol = state->mixer->protocol;
+	struct uac_input_terminal_descriptor *d = p1;
+
+	term->type = le16_to_cpu(d->wTerminalType);
+	term->channels = d->bNrChannels;
+	term->chconfig = le16_to_cpu(d->wChannelConfig);
+	term->name = d->iTerminal;
+	return 0;
+}
+
+static int parse_term_uac2_iterm_unit(struct mixer_build *state,
+				      struct usb_audio_term *term,
+				      void *p1, int id)
+{
+	struct uac2_input_terminal_descriptor *d = p1;
 	int err;
-	void *p1;
-	unsigned char *hdr;
 
-	memset(term, 0, sizeof(*term));
-	for (;;) {
-		/* a loop in the terminal chain? */
-		if (test_and_set_bit(id, state->termbitmap))
-			return -EINVAL;
+	/* call recursively to verify the referenced clock entity */
+	err = __check_input_term(state, d->bCSourceID, term);
+	if (err < 0)
+		return err;
 
-		p1 = find_audio_control_unit(state, id);
-		if (!p1)
-			break;
-		if (!snd_usb_validate_audio_desc(p1, protocol))
-			break; /* bad descriptor */
+	/* save input term properties after recursion,
+	 * to ensure they are not overriden by the recursion calls
+	 */
+	term->id = id;
+	term->type = le16_to_cpu(d->wTerminalType);
+	term->channels = d->bNrChannels;
+	term->chconfig = le32_to_cpu(d->bmChannelConfig);
+	term->name = d->iTerminal;
+	return 0;
+}
 
-		hdr = p1;
-		term->id = id;
+static int parse_term_uac3_iterm_unit(struct mixer_build *state,
+				      struct usb_audio_term *term,
+				      void *p1, int id)
+{
+	struct uac3_input_terminal_descriptor *d = p1;
+	int err;
 
-		if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
-			switch (hdr[2]) {
-			case UAC_INPUT_TERMINAL:
-				if (protocol == UAC_VERSION_1) {
-					struct uac_input_terminal_descriptor *d = p1;
-
-					term->type = le16_to_cpu(d->wTerminalType);
-					term->channels = d->bNrChannels;
-					term->chconfig = le16_to_cpu(d->wChannelConfig);
-					term->name = d->iTerminal;
-				} else { /* UAC_VERSION_2 */
-					struct uac2_input_terminal_descriptor *d = p1;
-
-					/* call recursively to verify that the
-					 * referenced clock entity is valid */
-					err = __check_input_term(state, d->bCSourceID, term);
-					if (err < 0)
-						return err;
+	/* call recursively to verify the referenced clock entity */
+	err = __check_input_term(state, d->bCSourceID, term);
+	if (err < 0)
+		return err;
 
-					/* save input term properties after recursion,
-					 * to ensure they are not overriden by the
-					 * recursion calls */
-					term->id = id;
-					term->type = le16_to_cpu(d->wTerminalType);
-					term->channels = d->bNrChannels;
-					term->chconfig = le32_to_cpu(d->bmChannelConfig);
-					term->name = d->iTerminal;
-				}
-				return 0;
-			case UAC_FEATURE_UNIT: {
-				/* the header is the same for v1 and v2 */
-				struct uac_feature_unit_descriptor *d = p1;
+	/* save input term properties after recursion,
+	 * to ensure they are not overriden by the recursion calls
+	 */
+	term->id = id;
+	term->type = le16_to_cpu(d->wTerminalType);
 
-				id = d->bSourceID;
-				break; /* continue to parse */
-			}
-			case UAC_MIXER_UNIT: {
-				struct uac_mixer_unit_descriptor *d = p1;
-
-				term->type = UAC3_MIXER_UNIT << 16; /* virtual type */
-				term->channels = uac_mixer_unit_bNrChannels(d);
-				term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol);
-				term->name = uac_mixer_unit_iMixer(d);
-				return 0;
-			}
-			case UAC_SELECTOR_UNIT:
-			case UAC2_CLOCK_SELECTOR: {
-				struct uac_selector_unit_descriptor *d = p1;
-				/* call recursively to retrieve the channel info */
-				err = __check_input_term(state, d->baSourceID[0], term);
-				if (err < 0)
-					return err;
-				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
-				term->id = id;
-				term->name = uac_selector_unit_iSelector(d);
-				return 0;
-			}
-			case UAC1_PROCESSING_UNIT:
-			/* UAC2_EFFECT_UNIT */
-				if (protocol == UAC_VERSION_1)
-					term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */
-				else /* UAC_VERSION_2 */
-					term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */
-				/* fall through */
-			case UAC1_EXTENSION_UNIT:
-			/* UAC2_PROCESSING_UNIT_V2 */
-				if (protocol == UAC_VERSION_1 && !term->type)
-					term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */
-				else if (protocol == UAC_VERSION_2 && !term->type)
-					term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */
-				/* fall through */
-			case UAC2_EXTENSION_UNIT_V2: {
-				struct uac_processing_unit_descriptor *d = p1;
-
-				if (protocol == UAC_VERSION_2 &&
-					hdr[2] == UAC2_EFFECT_UNIT) {
-					/* UAC2/UAC1 unit IDs overlap here in an
-					 * uncompatible way. Ignore this unit for now.
-					 */
-					return 0;
-				}
+	err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID));
+	if (err < 0)
+		return err;
+	term->channels = err;
 
-				if (d->bNrInPins) {
-					id = d->baSourceID[0];
-					break; /* continue to parse */
-				}
-				if (!term->type)
-					term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */
+	/* REVISIT: UAC3 IT doesn't have channels cfg */
+	term->chconfig = 0;
 
-				term->channels = uac_processing_unit_bNrChannels(d);
-				term->chconfig = uac_processing_unit_wChannelConfig(d, protocol);
-				term->name = uac_processing_unit_iProcessing(d, protocol);
-				return 0;
-			}
-			case UAC2_CLOCK_SOURCE: {
-				struct uac_clock_source_descriptor *d = p1;
+	term->name = le16_to_cpu(d->wTerminalDescrStr);
+	return 0;
+}
 
-				term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */
-				term->id = id;
-				term->name = d->iClockSource;
-				return 0;
-			}
-			default:
-				return -ENODEV;
-			}
-		} else { /* UAC_VERSION_3 */
-			switch (hdr[2]) {
-			case UAC_INPUT_TERMINAL: {
-				struct uac3_input_terminal_descriptor *d = p1;
-
-				/* call recursively to verify that the
-				 * referenced clock entity is valid */
-				err = __check_input_term(state, d->bCSourceID, term);
-				if (err < 0)
-					return err;
+static int parse_term_mixer_unit(struct mixer_build *state,
+				 struct usb_audio_term *term,
+				 void *p1, int id)
+{
+	struct uac_mixer_unit_descriptor *d = p1;
+	int protocol = state->mixer->protocol;
+	int err;
 
-				/* save input term properties after recursion,
-				 * to ensure they are not overriden by the
-				 * recursion calls */
-				term->id = id;
-				term->type = le16_to_cpu(d->wTerminalType);
+	err = uac_mixer_unit_get_channels(state, d);
+	if (err <= 0)
+		return err;
 
-				err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID));
-				if (err < 0)
-					return err;
-				term->channels = err;
+	term->type = UAC3_MIXER_UNIT << 16; /* virtual type */
+	term->channels = err;
+	if (protocol != UAC_VERSION_3) {
+		term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol);
+		term->name = uac_mixer_unit_iMixer(d);
+	}
+	return 0;
+}
+
+static int parse_term_selector_unit(struct mixer_build *state,
+				    struct usb_audio_term *term,
+				    void *p1, int id)
+{
+	struct uac_selector_unit_descriptor *d = p1;
+	int err;
 
-				/* REVISIT: UAC3 IT doesn't have channels cfg */
-				term->chconfig = 0;
+	/* call recursively to retrieve the channel info */
+	err = __check_input_term(state, d->baSourceID[0], term);
+	if (err < 0)
+		return err;
+	term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
+	term->id = id;
+	if (state->mixer->protocol != UAC_VERSION_3)
+		term->name = uac_selector_unit_iSelector(d);
+	return 0;
+}
 
-				term->name = le16_to_cpu(d->wTerminalDescrStr);
-				return 0;
-			}
-			case UAC3_FEATURE_UNIT: {
-				struct uac3_feature_unit_descriptor *d = p1;
+static int parse_term_proc_unit(struct mixer_build *state,
+				struct usb_audio_term *term,
+				void *p1, int id, int vtype)
+{
+	struct uac_processing_unit_descriptor *d = p1;
+	int protocol = state->mixer->protocol;
+	int err;
 
-				id = d->bSourceID;
-				break; /* continue to parse */
-			}
-			case UAC3_CLOCK_SOURCE: {
-				struct uac3_clock_source_descriptor *d = p1;
+	if (d->bNrInPins) {
+		/* call recursively to retrieve the channel info */
+		err = __check_input_term(state, d->baSourceID[0], term);
+		if (err < 0)
+			return err;
+	}
 
-				term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */
-				term->id = id;
-				term->name = le16_to_cpu(d->wClockSourceStr);
-				return 0;
-			}
-			case UAC3_MIXER_UNIT: {
-				struct uac_mixer_unit_descriptor *d = p1;
+	term->type = vtype << 16; /* virtual type */
+	term->id = id;
 
-				err = uac_mixer_unit_get_channels(state, d);
-				if (err <= 0)
-					return err;
+	if (protocol == UAC_VERSION_3)
+		return 0;
 
-				term->channels = err;
-				term->type = UAC3_MIXER_UNIT << 16; /* virtual type */
+	if (!term->channels) {
+		term->channels = uac_processing_unit_bNrChannels(d);
+		term->chconfig = uac_processing_unit_wChannelConfig(d, protocol);
+	}
+	term->name = uac_processing_unit_iProcessing(d, protocol);
+	return 0;
+}
 
-				return 0;
-			}
-			case UAC3_SELECTOR_UNIT:
-			case UAC3_CLOCK_SELECTOR: {
-				struct uac_selector_unit_descriptor *d = p1;
-				/* call recursively to retrieve the channel info */
-				err = __check_input_term(state, d->baSourceID[0], term);
-				if (err < 0)
-					return err;
-				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
-				term->id = id;
-				term->name = 0; /* TODO: UAC3 Class-specific strings */
+static int parse_term_uac2_clock_source(struct mixer_build *state,
+					struct usb_audio_term *term,
+					void *p1, int id)
+{
+	struct uac_clock_source_descriptor *d = p1;
 
-				return 0;
-			}
-			case UAC3_PROCESSING_UNIT: {
-				struct uac_processing_unit_descriptor *d = p1;
+	term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */
+	term->id = id;
+	term->name = d->iClockSource;
+	return 0;
+}
 
-				if (!d->bNrInPins)
-					return -EINVAL;
+static int parse_term_uac3_clock_source(struct mixer_build *state,
+					struct usb_audio_term *term,
+					void *p1, int id)
+{
+	struct uac3_clock_source_descriptor *d = p1;
 
-				/* call recursively to retrieve the channel info */
-				err = __check_input_term(state, d->baSourceID[0], term);
-				if (err < 0)
-					return err;
+	term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */
+	term->id = id;
+	term->name = le16_to_cpu(d->wClockSourceStr);
+	return 0;
+}
 
-				term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */
-				term->id = id;
-				term->name = 0; /* TODO: UAC3 Class-specific strings */
+#define PTYPE(a, b)	((a) << 8 | (b))
 
-				return 0;
-			}
-			default:
-				return -ENODEV;
-			}
+/*
+ * parse the source unit recursively until it reaches to a terminal
+ * or a branched unit.
+ */
+static int __check_input_term(struct mixer_build *state, int id,
+			      struct usb_audio_term *term)
+{
+	int protocol = state->mixer->protocol;
+	void *p1;
+	unsigned char *hdr;
+
+	for (;;) {
+		/* a loop in the terminal chain? */
+		if (test_and_set_bit(id, state->termbitmap))
+			return -EINVAL;
+
+		p1 = find_audio_control_unit(state, id);
+		if (!p1)
+			break;
+		if (!snd_usb_validate_audio_desc(p1, protocol))
+			break; /* bad descriptor */
+
+		hdr = p1;
+		term->id = id;
+
+		switch (PTYPE(protocol, hdr[2])) {
+		case PTYPE(UAC_VERSION_1, UAC_FEATURE_UNIT):
+		case PTYPE(UAC_VERSION_2, UAC_FEATURE_UNIT):
+		case PTYPE(UAC_VERSION_3, UAC3_FEATURE_UNIT): {
+			/* the header is the same for all versions */
+			struct uac_feature_unit_descriptor *d = p1;
+
+			id = d->bSourceID;
+			break; /* continue to parse */
+		}
+		case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL):
+			return parse_term_uac1_iterm_unit(state, term, p1, id);
+		case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL):
+			return parse_term_uac2_iterm_unit(state, term, p1, id);
+		case PTYPE(UAC_VERSION_3, UAC_INPUT_TERMINAL):
+			return parse_term_uac3_iterm_unit(state, term, p1, id);
+		case PTYPE(UAC_VERSION_1, UAC_MIXER_UNIT):
+		case PTYPE(UAC_VERSION_2, UAC_MIXER_UNIT):
+		case PTYPE(UAC_VERSION_3, UAC3_MIXER_UNIT):
+			return parse_term_mixer_unit(state, term, p1, id);
+		case PTYPE(UAC_VERSION_1, UAC_SELECTOR_UNIT):
+		case PTYPE(UAC_VERSION_2, UAC_SELECTOR_UNIT):
+		case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SELECTOR):
+		case PTYPE(UAC_VERSION_3, UAC3_SELECTOR_UNIT):
+		case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SELECTOR):
+			return parse_term_selector_unit(state, term, p1, id);
+		case PTYPE(UAC_VERSION_1, UAC1_PROCESSING_UNIT):
+		case PTYPE(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2):
+		case PTYPE(UAC_VERSION_3, UAC3_PROCESSING_UNIT):
+			return parse_term_proc_unit(state, term, p1, id,
+						    UAC3_PROCESSING_UNIT);
+		case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT):
+		case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT):
+			return parse_term_proc_unit(state, term, p1, id,
+						    UAC3_EFFECT_UNIT);
+		case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT):
+		case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2):
+		case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT):
+			return parse_term_proc_unit(state, term, p1, id,
+						    UAC3_EXTENSION_UNIT);
+		case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SOURCE):
+			return parse_term_uac2_clock_source(state, term, p1, id);
+		case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SOURCE):
+			return parse_term_uac3_clock_source(state, term, p1, id);
+		default:
+			return -ENODEV;
 		}
 	}
 	return -ENODEV;
@@ -2712,7 +2730,6 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 		return 0; /* skip invalid unit */
 	}
 
-#define PTYPE(a, b)	((a) << 8 | (b))
 	switch (PTYPE(protocol, p1[2])) {
 	case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL):
 	case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL):
-- 
cgit v1.2.3-59-g8ed1b


From 60849562a5db4a1eee2160167e4dce4590d3eafe Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 26 Aug 2019 13:55:21 +0200
Subject: ALSA: usb-audio: Fix possible NULL dereference at
 create_yamaha_midi_quirk()

The previous addition of descriptor validation may lead to a NULL
dereference at create_yamaha_midi_quirk() when either injd or outjd is
NULL.  Add proper non-NULL checks.

Fixes: 57f8770620e9 ("ALSA: usb-audio: More validations of descriptor units")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 7e9735aa7ac9..5c0fa5cf1987 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -248,8 +248,8 @@ static int create_yamaha_midi_quirk(struct snd_usb_audio *chip,
 					NULL, USB_MS_MIDI_OUT_JACK);
 	if (!injd && !outjd)
 		return -ENODEV;
-	if (!snd_usb_validate_midi_desc(injd) ||
-	    !snd_usb_validate_midi_desc(outjd))
+	if (!(injd && snd_usb_validate_midi_desc(injd)) ||
+	    !(outjd && snd_usb_validate_midi_desc(outjd)))
 		return -ENODEV;
 	if (injd && (injd->bLength < 5 ||
 		     (injd->bJackType != USB_MS_EMBEDDED &&
-- 
cgit v1.2.3-59-g8ed1b


From b39e077fcb283dd96dd251a3abeba585402c61fe Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Mon, 26 Aug 2019 16:45:50 +0300
Subject: ALSA: usb-audio: remove some dead code

We recently cleaned up the error handling in commit 52c3e317a857 ("ALSA:
usb-audio: Unify the release of usb_mixer_elem_info objects") but
accidentally left this stray return.

Fixes: 52c3e317a857 ("ALSA: usb-audio: Unify the release of usb_mixer_elem_info objects")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 83d22f241bdd..3fd1d1749edf 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2651,7 +2651,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 		usb_audio_err(state->chip, "cannot malloc kcontrol\n");
 		err = -ENOMEM;
 		goto error_name;
-		return -ENOMEM;
 	}
 	kctl->private_value = (unsigned long)namelist;
 	kctl->private_free = usb_mixer_selector_elem_free;
-- 
cgit v1.2.3-59-g8ed1b


From 4c098dab121a4164e9daa4d1ddbc59e93054e79e Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 26 Aug 2019 22:54:15 +0900
Subject: ALSA: oxfw: fix NULL pointer dereference to unused stream structure

ALSA oxfw driver supports Griffin FireWave. This device supports one
isochronous stream for PCM playback. Current driver executes code to
wait event for stream structure of opposite direction. This causes
NULL pointer dereference.

This commit fixes the bug.

Fixes: ac5d77864cfc ("ALSA: oxfw: support AMDTP domain")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/oxfw/oxfw-stream.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 0318dc4dcc55..3c9a796b6526 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -350,13 +350,18 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw)
 
 		// Wait first packet.
 		if (!amdtp_stream_wait_callback(&oxfw->rx_stream,
-						CALLBACK_TIMEOUT) ||
-		    !amdtp_stream_wait_callback(&oxfw->tx_stream,
 						CALLBACK_TIMEOUT)) {
 			err = -ETIMEDOUT;
 			goto error;
 		}
 
+		if (oxfw->has_output) {
+			if (!amdtp_stream_wait_callback(&oxfw->tx_stream,
+							CALLBACK_TIMEOUT)) {
+				err = -ETIMEDOUT;
+				goto error;
+			}
+		}
 	}
 
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From b9a4805a4ac1200326ee0c78df0ae3faceabad49 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Fri, 23 Aug 2019 20:59:39 +0800
Subject: ASoC: SOF: imx8: Make some functions static

Fix sparse warnings:

sound/soc/sof/imx/imx8.c:104:6: warning: symbol 'imx8_dsp_handle_reply' was not declared. Should it be static?
sound/soc/sof/imx/imx8.c:115:6: warning: symbol 'imx8_dsp_handle_request' was not declared. Should it be static?
sound/soc/sof/imx/imx8.c:336:5: warning: symbol 'imx8_get_bar_index' was not declared. Should it be static?
sound/soc/sof/imx/imx8.c:341:6: warning: symbol 'imx8_ipc_msg_data' was not declared. Should it be static?
sound/soc/sof/imx/imx8.c:348:5: warning: symbol 'imx8_ipc_pcm_params' was not declared. Should it be static?

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20190823125939.30012-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/imx/imx8.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index e502f584207f..640472491037 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -101,7 +101,7 @@ static int imx8_get_window_offset(struct snd_sof_dev *sdev, u32 id)
 	return MBOX_OFFSET;
 }
 
-void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc)
+static void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc)
 {
 	struct imx8_priv *priv = imx_dsp_get_data(ipc);
 	unsigned long flags;
@@ -112,7 +112,7 @@ void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc)
 	spin_unlock_irqrestore(&priv->sdev->ipc_lock, flags);
 }
 
-void imx8_dsp_handle_request(struct imx_dsp_ipc *ipc)
+static void imx8_dsp_handle_request(struct imx_dsp_ipc *ipc)
 {
 	struct imx8_priv *priv = imx_dsp_get_data(ipc);
 
@@ -333,21 +333,21 @@ static int imx8_remove(struct snd_sof_dev *sdev)
 }
 
 /* on i.MX8 there is 1 to 1 match between type and BAR idx */
-int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type)
+static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 {
 	return type;
 }
 
-void imx8_ipc_msg_data(struct snd_sof_dev *sdev,
-		       struct snd_pcm_substream *substream,
-		       void *p, size_t sz)
+static void imx8_ipc_msg_data(struct snd_sof_dev *sdev,
+			      struct snd_pcm_substream *substream,
+			      void *p, size_t sz)
 {
 	sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz);
 }
 
-int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
-			struct snd_pcm_substream *substream,
-			const struct sof_ipc_pcm_params_reply *reply)
+static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
+			       struct snd_pcm_substream *substream,
+			       const struct sof_ipc_pcm_params_reply *reply)
 {
 	return 0;
 }
-- 
cgit v1.2.3-59-g8ed1b


From 9573820eb1951e0cb0f329886abcb4153f2ea798 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Tue, 27 Aug 2019 11:55:15 -0400
Subject: ASoC: imx-audmix: register the card on a proper dev

This platform device is registered from "fsl_audmix", which is
its parent device. If use pdev->dev.parent for the priv->card.dev,
the value set by dev_set_drvdata in parent device will be covered
by the value in child device.

Fixes: b86ef5367761 ("ASoC: fsl: Add Audio Mixer machine driver")
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/1566921315-23402-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/imx-audmix.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 9e1cb18859ce..71590ca6394b 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -325,14 +325,14 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	priv->card.num_configs = priv->num_dai_conf;
 	priv->card.dapm_routes = priv->dapm_routes;
 	priv->card.num_dapm_routes = priv->num_dapm_routes;
-	priv->card.dev = pdev->dev.parent;
+	priv->card.dev = &pdev->dev;
 	priv->card.owner = THIS_MODULE;
 	priv->card.name = "imx-audmix";
 
 	platform_set_drvdata(pdev, &priv->card);
 	snd_soc_card_set_drvdata(&priv->card, priv);
 
-	ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card);
+	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed\n");
 		return ret;
-- 
cgit v1.2.3-59-g8ed1b


From 455b1d42e82c6027b9763f0055b54e45ff6cd7fd Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Tue, 27 Aug 2019 11:32:05 +0200
Subject: ASoC: sunxi: Revert initial A83t support

This reverts commit 3e9acd7ac6933c (ASoC: sun4i-i2s: Remove
duplicated quirks structure").

It turns out that while one I2S controller is described in the A83t
datasheet, the driver supports another, undocumented, one that has been
inherited from the older SoCs, while the documented one uses the new
design.

Fixes: 3e9acd7ac693 ("ASoC: sun4i-i2s: Remove duplicated quirks structure")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/20190827093206.17919-1-mripard@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 25 ++++++++++++++++++++++++-
 1 file changed, 24 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 57bf2a33753e..a6a3f772fdf0 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1097,6 +1097,11 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = {
 	.set_fmt		= sun4i_i2s_set_soc_fmt,
 };
 
+/*
+ * This doesn't describe the TDM controller documented in the A83t
+ * datasheet, but the three undocumented I2S controller that use the
+ * older design.
+ */
 static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
@@ -1115,6 +1120,24 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.set_fmt		= sun8i_i2s_set_soc_fmt,
 };
 
+static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
+	.has_reset		= true,
+	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
+	.sun4i_i2s_regmap	= &sun8i_i2s_regmap_config,
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
+	.bclk_dividers		= sun8i_i2s_clk_div,
+	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
+	.mclk_dividers		= sun8i_i2s_clk_div,
+	.num_mclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
+	.get_bclk_parent_rate	= sun8i_i2s_get_bclk_parent_rate,
+	.get_sr			= sun8i_i2s_get_sr_wss,
+	.get_wss		= sun8i_i2s_get_sr_wss,
+	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
+	.set_fmt		= sun8i_i2s_set_soc_fmt,
+};
+
 static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
@@ -1296,7 +1319,7 @@ static const struct of_device_id sun4i_i2s_match[] = {
 	},
 	{
 		.compatible = "allwinner,sun8i-h3-i2s",
-		.data = &sun8i_a83t_i2s_quirks,
+		.data = &sun8i_h3_i2s_quirks,
 	},
 	{
 		.compatible = "allwinner,sun50i-a64-codec-i2s",
-- 
cgit v1.2.3-59-g8ed1b


From dd23e1d566d0f74aa3b68ab3237927bb15f0e644 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 27 Aug 2019 16:37:50 +0200
Subject: ALSA: hda - Allow runtime PM for controller if component notifier is
 used

Currently we disallow the runtime PM of the HD-audio controller if
it's bound with HDMI/DP on Nvidia / AMD unless it's for dGPU.  This is
for keeping the link up to get the proper notification for ELD
hotplug.

As explained in the commit 37a3a98ef601 ("ALSA: hda - Enable runtime
PM only for discrete GPU"), this keep-power-up behavior is rather a
stop-gap solution until the ELD notification via audio component.
And now we finally got the audio component for these graphics drivers
via commit ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for
AMD/ATI and Nvidia HDMI"), so it's time to change.

This patch makes HD-audio controller again runtime-suspendable when
the device gets bound with audio component in HDMI codec driver.  For
making it easier to access from the codec driver, move the flag into
the common hda_bus object instead of hda_intel flag.  Also rename it
to keep_power, to indicate the actual meaning.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hda_codec.h  |  1 +
 sound/pci/hda/hda_intel.c  | 11 +++++------
 sound/pci/hda/hda_intel.h  |  1 -
 sound/pci/hda/patch_hdmi.c |  4 +++-
 4 files changed, 9 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index 871993696c5f..9a0393cf024c 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -59,6 +59,7 @@ struct hda_bus {
 	unsigned int in_reset:1;	/* during reset operation */
 	unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
 	unsigned int bus_probing :1;	/* during probing process */
+	unsigned int keep_power:1;	/* keep power up for notification */
 
 	int primary_dig_out_type;	/* primary digital out PCM type */
 	unsigned int mixer_assigned;	/* codec addr for mixer name */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4496fce21300..91e71be42fa4 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -356,7 +356,7 @@ enum {
  */
 #ifdef SUPPORT_VGA_SWITCHEROO
 #define use_vga_switcheroo(chip)	((chip)->use_vga_switcheroo)
-#define needs_eld_notify_link(chip)	((chip)->need_eld_notify_link)
+#define needs_eld_notify_link(chip)	((chip)->bus.keep_power)
 #else
 #define use_vga_switcheroo(chip)	0
 #define needs_eld_notify_link(chip)	false
@@ -1145,7 +1145,7 @@ static int azx_runtime_idle(struct device *dev)
 		return -EBUSY;
 
 	/* ELD notification gets broken when HD-audio bus is off */
-	if (needs_eld_notify_link(hda))
+	if (needs_eld_notify_link(chip))
 		return -EBUSY;
 
 	return 0;
@@ -1256,7 +1256,7 @@ static void setup_vga_switcheroo_runtime_pm(struct azx *chip)
 	struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
 	struct hda_codec *codec;
 
-	if (hda->use_vga_switcheroo && !hda->need_eld_notify_link) {
+	if (hda->use_vga_switcheroo && !needs_eld_notify_link(chip)) {
 		list_for_each_codec(codec, &chip->bus)
 			codec->auto_runtime_pm = 1;
 		/* reset the power save setup */
@@ -1270,10 +1270,9 @@ static void azx_vs_gpu_bound(struct pci_dev *pci,
 {
 	struct snd_card *card = pci_get_drvdata(pci);
 	struct azx *chip = card->private_data;
-	struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
 
 	if (client_id == VGA_SWITCHEROO_DIS)
-		hda->need_eld_notify_link = 0;
+		chip->bus.keep_power = 0;
 	setup_vga_switcheroo_runtime_pm(chip);
 }
 
@@ -1285,7 +1284,7 @@ static void init_vga_switcheroo(struct azx *chip)
 		dev_info(chip->card->dev,
 			 "Handle vga_switcheroo audio client\n");
 		hda->use_vga_switcheroo = 1;
-		hda->need_eld_notify_link = 1; /* cleared in gpu_bound op */
+		chip->bus.keep_power = 1; /* cleared in either gpu_bound op or codec probe */
 		chip->driver_caps |= AZX_DCAPS_PM_RUNTIME;
 		pci_dev_put(p);
 	}
diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h
index 1468865e0342..2acfff3da1a0 100644
--- a/sound/pci/hda/hda_intel.h
+++ b/sound/pci/hda/hda_intel.h
@@ -25,7 +25,6 @@ struct hda_intel {
 
 	/* vga_switcheroo setup */
 	unsigned int use_vga_switcheroo:1;
-	unsigned int need_eld_notify_link:1;
 	unsigned int vga_switcheroo_registered:1;
 	unsigned int init_failed:1; /* delayed init failed */
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 933c7bf47ef6..83b8b9d27711 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2542,8 +2542,10 @@ static void generic_acomp_init(struct hda_codec *codec,
 	spec->port2pin = port2pin;
 	setup_drm_audio_ops(codec, ops);
 	if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
-				 match_bound_vga, 0))
+				 match_bound_vga, 0)) {
 		spec->acomp_registered = true;
+		codec->bus->keep_power = 0;
+	}
 }
 
 /*
-- 
cgit v1.2.3-59-g8ed1b


From 3cdc3c9fd9fa92bcef777512d9f4eef854f9a4ed Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Mon, 26 Aug 2019 11:26:42 +0800
Subject: ASoC: mediatek: mt8183: remove unused DAPM-related variables

Remove unused variable mt8183_mt6358_ts3a227_max98357_dapm_widgets and
mt8183_mt6358_ts3a227_max98357_dapm_routes.  They are accidentially
included when rebasing commits.

Fixes: 6191cbde5ff0 ("ASoC: mediatek: mt8183: switch tdm pins gpio function when playback on or off")
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190826032642.27324-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 10 ----------
 1 file changed, 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index fe8a7461ace5..951706fa75fc 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -46,16 +46,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static const struct snd_soc_dapm_widget
-mt8183_mt6358_ts3a227_max98357_dapm_widgets[] = {
-	SND_SOC_DAPM_OUTPUT("IT6505_8CH"),
-};
-
-static const struct snd_soc_dapm_route
-mt8183_mt6358_ts3a227_max98357_dapm_routes[] = {
-	{"IT6505_8CH", NULL, "TDM"},
-};
-
 enum PINCTRL_PIN_STATE {
 	PIN_STATE_DEFAULT = 0,
 	PIN_TDM_OUT_ON,
-- 
cgit v1.2.3-59-g8ed1b


From 393151c211006cc5ac6af26ecd9982dd916a8104 Mon Sep 17 00:00:00 2001
From: Wei Yongjun <weiyongjun1@huawei.com>
Date: Mon, 26 Aug 2019 12:00:03 +0000
Subject: ASoC: SOF: imx8: Fix return value check in imx8_probe()

In case of error, the function devm_ioremap_wc() returns NULL pointer
not ERR_PTR(). The IS_ERR() test in the return value check should be
replaced with NULL test.

Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20190826120003.183279-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/imx/imx8.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 640472491037..c9d849ced54a 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -296,10 +296,10 @@ static int imx8_probe(struct snd_sof_dev *sdev)
 	sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, res.start,
 							  res.end - res.start +
 							  1);
-	if (IS_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM])) {
+	if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) {
 		dev_err(sdev->dev, "failed to ioremap mem 0x%x size 0x%x\n",
 			base, size);
-		ret = PTR_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM]);
+		ret = -ENOMEM;
 		goto exit_pdev_unregister;
 	}
 	sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM;
-- 
cgit v1.2.3-59-g8ed1b


From 9ec05d4723bf83dd272cef5ccf508e5fe4d30fa3 Mon Sep 17 00:00:00 2001
From: Maxime Ripard <maxime.ripard@bootlin.com>
Date: Tue, 27 Aug 2019 14:31:31 +0200
Subject: ASoC: sun4i: Revert A83t description

The last set of reworks included some fixes to change the A83t behaviour
and "fix" it.

It turns out that the controller described in the datasheet and the one
supported here are not the same, yet the A83t has the two of them, and the
one supported in the driver wasn't the one described in the datasheet.

Fix this by reintroducing the proper quirks.

Fixes: 69e450e50ca6 ("ASoC: sun4i-i2s: Fix the LRCK period on A83t")
Fixes: bf943d527987 ("ASoC: sun4i-i2s: Fix MCLK Enable bit offset on A83t")
Fixes: 2e04fc4dbf50 ("ASoC: sun4i-i2s: Fix WSS and SR fields for the A83t")
Fixes: 515fcfbc7736 ("ASoC: sun4i-i2s: Fix LRCK and BCLK polarity offsets on newer SoCs")
Fixes: c1d3a921d72b ("ASoC: sun4i-i2s: Fix the MCLK and BCLK dividers on newer SoCs")
Fixes: fb19739d7f68 ("ASoC: sun4i-i2s: Use module clock as BCLK parent on newer SoCs")
Fixes: 71137bcd0a9a ("ASoC: sun4i-i2s: Move the format configuration to a callback")
Fixes: d70be625f25a ("ASoC: sun4i-i2s: Move the channel configuration to a callback")
Reported-by: Chen-Yu Tsai <wens@csie.org>
Tested-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/20190827123131.29129-2-mripard@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index a6a3f772fdf0..d0a8d5810c0a 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1106,18 +1106,18 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = {
 	.has_reset		= true,
 	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
 	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
-	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8),
-	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2),
-	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6),
-	.bclk_dividers		= sun8i_i2s_clk_div,
-	.num_bclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
-	.mclk_dividers		= sun8i_i2s_clk_div,
-	.num_mclk_dividers	= ARRAY_SIZE(sun8i_i2s_clk_div),
-	.get_bclk_parent_rate	= sun8i_i2s_get_bclk_parent_rate,
-	.get_sr			= sun8i_i2s_get_sr_wss,
-	.get_wss		= sun8i_i2s_get_sr_wss,
-	.set_chan_cfg		= sun8i_i2s_set_chan_cfg,
-	.set_fmt		= sun8i_i2s_set_soc_fmt,
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+	.bclk_dividers		= sun4i_i2s_bclk_div,
+	.num_bclk_dividers	= ARRAY_SIZE(sun4i_i2s_bclk_div),
+	.mclk_dividers		= sun4i_i2s_mclk_div,
+	.num_mclk_dividers	= ARRAY_SIZE(sun4i_i2s_mclk_div),
+	.get_bclk_parent_rate	= sun4i_i2s_get_bclk_parent_rate,
+	.get_sr			= sun4i_i2s_get_sr,
+	.get_wss		= sun4i_i2s_get_wss,
+	.set_chan_cfg		= sun4i_i2s_set_chan_cfg,
+	.set_fmt		= sun4i_i2s_set_soc_fmt,
 };
 
 static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
-- 
cgit v1.2.3-59-g8ed1b


From a325c7bc3a92f1d5ba8cff10d7ab93a989cdeb80 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Mon, 26 Aug 2019 16:18:55 +0300
Subject: ASoC: SOF: imx8: Fix an is IS_ERR() vs NULL check

The device_link_add() function only returns NULL on error, it doesn't
return error pointers.

Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20190826131855.GA6840@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/imx/imx8.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index c9d849ced54a..2a22b18e5ec0 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -227,8 +227,8 @@ static int imx8_probe(struct snd_sof_dev *sdev)
 						DL_FLAG_STATELESS |
 						DL_FLAG_PM_RUNTIME |
 						DL_FLAG_RPM_ACTIVE);
-		if (IS_ERR(priv->link[i])) {
-			ret = PTR_ERR(priv->link[i]);
+		if (!priv->link[i]) {
+			ret = -ENOMEM;
 			dev_pm_domain_detach(priv->pd_dev[i], false);
 			goto exit_unroll_pm;
 		}
-- 
cgit v1.2.3-59-g8ed1b


From 17d29ff98fd4b70e9ccdac5e95e18a087e2737ef Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Date: Tue, 27 Aug 2019 16:17:07 +0200
Subject: ASoC: Intel: Skylake: Use correct function to access iomem space
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

For copying from __iomem, we should use __ioread32_copy.

reported by sparse:
sound/soc/intel/skylake/skl-debug.c:437:34: warning: incorrect type in argument 1 (different address spaces)
sound/soc/intel/skylake/skl-debug.c:437:34:    expected void [noderef] <asn:2> *to
sound/soc/intel/skylake/skl-debug.c:437:34:    got unsigned char *
sound/soc/intel/skylake/skl-debug.c:437:51: warning: incorrect type in argument 2 (different address spaces)
sound/soc/intel/skylake/skl-debug.c:437:51:    expected void const *from
sound/soc/intel/skylake/skl-debug.c:437:51:    got void [noderef] <asn:2> *[assigned] fw_reg_addr

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Link: https://lore.kernel.org/r/20190827141712.21015-2-amadeuszx.slawinski@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
---
 sound/soc/intel/skylake/skl-debug.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index b9b4a72a4334..b28a9c2b0380 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -188,7 +188,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf,
 	memset(d->fw_read_buff, 0, FW_REG_BUF);
 
 	if (w0_stat_sz > 0)
-		__iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
+		__ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
 
 	for (offset = 0; offset < FW_REG_SIZE; offset += 16) {
 		ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
-- 
cgit v1.2.3-59-g8ed1b


From 810f3b860850148788fc1ed8a6f5f807199fed65 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Date: Tue, 27 Aug 2019 16:17:08 +0200
Subject: ASoC: Intel: Fix use of potentially uninitialized variable
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

If ipc->ops.reply_msg_match is NULL, we may end up using uninitialized
mask value.

reported by smatch:
sound/soc/intel/common/sst-ipc.c:266 sst_ipc_reply_find_msg() error: uninitialized symbol 'mask'.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Link: https://lore.kernel.org/r/20190827141712.21015-3-amadeuszx.slawinski@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
---
 sound/soc/intel/common/sst-ipc.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c
index ef5b66af1cd2..3a66121ee9bb 100644
--- a/sound/soc/intel/common/sst-ipc.c
+++ b/sound/soc/intel/common/sst-ipc.c
@@ -222,6 +222,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc,
 
 	if (ipc->ops.reply_msg_match != NULL)
 		header = ipc->ops.reply_msg_match(header, &mask);
+	else
+		mask = (u64)-1;
 
 	if (list_empty(&ipc->rx_list)) {
 		dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n",
-- 
cgit v1.2.3-59-g8ed1b


From db33f00d15a63d269e283bad3f6f61eb00d2bc9d Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Date: Tue, 27 Aug 2019 16:17:09 +0200
Subject: ASoC: dapm: Expose snd_soc_dapm_new_control_unlocked properly
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

We use snd_soc_dapm_new_control_unlocked for topology and have local
declaration, instead declare it properly in header like already declared
snd_soc_dapm_new_control.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Link: https://lore.kernel.org/r/20190827141712.21015-4-amadeuszx.slawinski@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dapm.h | 3 +++
 sound/soc/soc-topology.c | 6 ------
 2 files changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c00a0b8ade08..8a90816a6eb5 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -402,6 +402,9 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 struct snd_soc_dapm_widget *snd_soc_dapm_new_control(
 		struct snd_soc_dapm_context *dapm,
 		const struct snd_soc_dapm_widget *widget);
+struct snd_soc_dapm_widget *snd_soc_dapm_new_control_unlocked(
+		struct snd_soc_dapm_context *dapm,
+		const struct snd_soc_dapm_widget *widget);
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 				 struct snd_soc_dai *dai);
 int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index dc463f1a9e24..2eca85c04a3e 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -80,12 +80,6 @@ struct soc_tplg {
 
 static int soc_tplg_process_headers(struct soc_tplg *tplg);
 static void soc_tplg_complete(struct soc_tplg *tplg);
-struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
-			 const struct snd_soc_dapm_widget *widget);
-struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
-			 const struct snd_soc_dapm_widget *widget);
 static void soc_tplg_denum_remove_texts(struct soc_enum *se);
 static void soc_tplg_denum_remove_values(struct soc_enum *se);
 
-- 
cgit v1.2.3-59-g8ed1b


From 855a06da37a773fd073d51023ac9d07988c87da8 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Date: Tue, 27 Aug 2019 16:17:12 +0200
Subject: ASoC: Intel: NHLT: Fix debug print format
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

oem_table_id is 8 chars long, so we need to limit it, otherwise it
may print some unprintable characters into dmesg.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Link: https://lore.kernel.org/r/20190827141712.21015-7-amadeuszx.slawinski@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
---
 sound/soc/intel/skylake/skl-nhlt.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 1132109cb992..e01815cec6fd 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -225,7 +225,7 @@ int skl_nhlt_update_topology_bin(struct skl *skl)
 	struct hdac_bus *bus = skl_to_bus(skl);
 	struct device *dev = bus->dev;
 
-	dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n",
+	dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n",
 		nhlt->header.oem_id, nhlt->header.oem_table_id,
 		nhlt->header.oem_revision);
 
-- 
cgit v1.2.3-59-g8ed1b


From 0cc33cecb4a1de5b55d382b8e24a128a3551a031 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Date: Tue, 27 Aug 2019 16:17:10 +0200
Subject: ASoC: Intel: Skylake: Print module type instead of id
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

When we are printing module params, we were actually printing module id
instead of type, but debug message was saying that number we get is type.
So print module type, as it is useful when debugging paths, but also
keep printing module id, as it is used in all other logs.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Link: https://lore.kernel.org/r/20190827141712.21015-5-amadeuszx.slawinski@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-messages.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index d43496c5f29e..476ef1897961 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -867,8 +867,9 @@ static int skl_set_module_format(struct skl_dev *skl,
 
 	}
 
-	dev_dbg(skl->dev, "Module type=%d config size: %d bytes\n",
-			module_config->id.module_id, param_size);
+	dev_dbg(skl->dev, "Module type=%d id=%d config size: %d bytes\n",
+			module_config->m_type, module_config->id.module_id,
+			param_size);
 	print_hex_dump_debug("Module params:", DUMP_PREFIX_OFFSET, 8, 4,
 			*param_data, param_size, false);
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From 6f43791713c5bb1212ec8497034275dfdbba5cbd Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Tue, 27 Aug 2019 16:17:11 +0200
Subject: ASoC: Intel: Skylake: Release topology when we are done with it
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Currently topology is kept in memory while driver is running. It's
unnecessary, as it's only needed during parsing.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20190827141712.21015-6-amadeuszx.slawinski@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/skylake/skl-topology.c | 20 ++++++++++----------
 sound/soc/intel/skylake/skl.h          |  1 -
 2 files changed, 10 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index ae5c75d03fdc..69cd7a81bf2a 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -3579,23 +3579,25 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus)
 	 * The complete tplg for SKL is loaded as index 0, we don't use
 	 * any other index
 	 */
-	ret = snd_soc_tplg_component_load(component,
-					&skl_tplg_ops, fw, 0);
+	ret = snd_soc_tplg_component_load(component, &skl_tplg_ops, fw, 0);
 	if (ret < 0) {
 		dev_err(bus->dev, "tplg component load failed%d\n", ret);
-		release_firmware(fw);
-		return -EINVAL;
+		goto err;
 	}
 
-	skl->tplg = fw;
 	ret = skl_tplg_create_pipe_widget_list(component);
-	if (ret < 0)
-		return ret;
+	if (ret < 0) {
+		dev_err(bus->dev, "tplg create pipe widget list failed%d\n",
+				ret);
+		goto err;
+	}
 
 	list_for_each_entry(ppl, &skl->ppl_list, node)
 		skl_tplg_set_pipe_type(skl, ppl->pipe);
 
-	return 0;
+err:
+	release_firmware(fw);
+	return ret;
 }
 
 void skl_tplg_exit(struct snd_soc_component *component, struct hdac_bus *bus)
@@ -3609,6 +3611,4 @@ void skl_tplg_exit(struct snd_soc_component *component, struct hdac_bus *bus)
 
 	/* clean up topology */
 	snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
-
-	release_firmware(skl->tplg);
 }
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index f8c714153610..2bfbf59277c4 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -75,7 +75,6 @@ struct skl_dev {
 	const char *fw_name;
 	char tplg_name[64];
 	unsigned short pci_id;
-	const struct firmware *tplg;
 
 	int supend_active;
 
-- 
cgit v1.2.3-59-g8ed1b


From f972d02fee2496024cfd6f59021c9d89d54922a6 Mon Sep 17 00:00:00 2001
From: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Date: Tue, 27 Aug 2019 00:38:59 +0900
Subject: ASoC: es8316: fix headphone mixer volume table

This patch fix setting table of Headphone mixer volume.
Current code uses 4 ... 7 values but these values are prohibited.

Correct settings are the following:
  0000 -12dB
  0001 -10.5dB
  0010 -9dB
  0011 -7.5dB
  0100 -6dB
  1000 -4.5dB
  1001 -3dB
  1010 -1.5dB
  1011 0dB

Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Reviewed-by: Daniel Drake <drake@endlessm.com>
Link: https://lore.kernel.org/r/20190826153900.25969-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8316.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 6db002cc2058..96d04896193f 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -51,7 +51,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0);
 static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0);
-static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv,
+	0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0),
+	8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0),
+);
 
 static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
 	0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0),
@@ -89,7 +92,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 	SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL,
 		       4, 0, 3, 1, hpout_vol_tlv),
 	SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL,
-		       0, 4, 7, 0, hpmixer_gain_tlv),
+		       0, 4, 11, 0, hpmixer_gain_tlv),
 
 	SOC_ENUM("Playback Polarity", dacpol),
 	SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
-- 
cgit v1.2.3-59-g8ed1b


From f6e77921969003eaf5dbae9c0b5feeb20c6caf50 Mon Sep 17 00:00:00 2001
From: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Date: Tue, 27 Aug 2019 00:39:00 +0900
Subject: ASoC: es8316: fix inverted L/R of headphone mixer volume

This patch fixes inverted Left-Right channel of headphone mixer
volume by wrong shift_left, shift_right values.

Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Reviewed-by: Daniel Drake <drake@endlessm.com>
Link: https://lore.kernel.org/r/20190826153900.25969-2-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8316.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 96d04896193f..ed2959dbe1fb 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -92,7 +92,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 	SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL,
 		       4, 0, 3, 1, hpout_vol_tlv),
 	SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL,
-		       0, 4, 11, 0, hpmixer_gain_tlv),
+		       4, 0, 11, 0, hpmixer_gain_tlv),
 
 	SOC_ENUM("Playback Polarity", dacpol),
 	SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
-- 
cgit v1.2.3-59-g8ed1b


From e9e897d5fdba9bda3a8c80ea39761413725f198c Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Mon, 26 Aug 2019 17:00:52 +0800
Subject: ASoC: rt1011: add mutex protection to set_fmt/set_tdm_slot

The calibration process at booting will reset registers and bypass cache
to make sure the calibration is done.
We add mutex protection to avoid unexpected settings while
the registration process and calibration are interleaved.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20190826090052.1875-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1011.c | 27 +++++++++++++++++++--------
 1 file changed, 19 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 0a6ff13d76e1..ed28250d5e34 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -1619,14 +1619,18 @@ static int rt1011_hw_params(struct snd_pcm_substream *substream,
 static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = dai->component;
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
 	unsigned int reg_val = 0, reg_bclk_inv = 0;
+	int ret = 0;
 
+	snd_soc_dapm_mutex_lock(dapm);
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
 		reg_val |= RT1011_I2S_TDM_MS_S;
 		break;
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -1636,7 +1640,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		reg_bclk_inv |= RT1011_TDM_INV_BCLK;
 		break;
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -1652,7 +1656,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		reg_val |= RT1011_I2S_TDM_DF_PCM_B;
 		break;
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
 	switch (dai->id) {
@@ -1667,9 +1671,11 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		break;
 	default:
 		dev_err(component->dev, "Invalid dai->id: %d\n", dai->id);
-		return -EINVAL;
+		ret = -EINVAL;
 	}
-	return 0;
+
+	snd_soc_dapm_mutex_unlock(dapm);
+	return ret;
 }
 
 static int rt1011_set_component_sysclk(struct snd_soc_component *component,
@@ -1788,8 +1794,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
 {
 	struct snd_soc_component *component = dai->component;
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
 	unsigned int val = 0, tdm_en = 0;
+	int ret = 0;
 
+	snd_soc_dapm_mutex_lock(dapm);
 	if (rx_mask || tx_mask)
 		tdm_en = RT1011_TDM_I2S_DOCK_EN_1;
 
@@ -1809,7 +1819,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai,
 	case 2:
 		break;
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
 	switch (slot_width) {
@@ -1828,7 +1838,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai,
 	case 16:
 		break;
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
 	snd_soc_component_update_bits(component, RT1011_TDM1_SET_1,
@@ -1845,7 +1855,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai,
 		RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG,
 		RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT);
 
-	return 0;
+	snd_soc_dapm_mutex_unlock(dapm);
+	return ret;
 }
 
 static int rt1011_probe(struct snd_soc_component *component)
-- 
cgit v1.2.3-59-g8ed1b


From d2ac1fe0342ce2b44621f60d053800e2d94f1d21 Mon Sep 17 00:00:00 2001
From: Jiaxin Yu <jiaxin.yu@mediatek.com>
Date: Sat, 24 Aug 2019 12:58:46 +0800
Subject: ASoC: mediatek: mt8183: fix tdm out data align issue

Mt8183 tdm out support S16_LE/S24_LE/S32_LE formats. When output S32_LE,
we need set hd_align so that memif can output MSB 24bits. When output
S24_LE, we need reset hd_align so that memif can output LSB 24bits.

Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1566622726-27113-1-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
index d16563408465..10ea4fdbeb1e 100644
--- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c
+++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
@@ -241,7 +241,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream,
 	struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
 	struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id];
 	int hd_audio = 0;
-	int hd_align = 1;
+	int hd_align = 0;
 
 	/* set hd mode */
 	switch (substream->runtime->format) {
@@ -254,7 +254,6 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream,
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		hd_audio = 1;
-		hd_align = 0;
 		break;
 	default:
 		dev_err(afe->dev, "%s() error: unsupported format %d\n",
-- 
cgit v1.2.3-59-g8ed1b


From b429ca49406501e2c48c2a137eab12910c21ad0c Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Tue, 27 Aug 2019 18:13:46 -0400
Subject: ASoC: cs42xx8: Force suspend/resume during system suspend/resume

Use force_suspend/resume to make sure clocks are disabled/enabled
accordingly during system suspend/resume.

Signed-off-by: Dong Aisheng <aisheng.dong@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/1566944026-18113-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42xx8.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 5b049fcdba20..94b1adb088fd 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -684,6 +684,8 @@ static int cs42xx8_runtime_suspend(struct device *dev)
 #endif
 
 const struct dev_pm_ops cs42xx8_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
 	SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL)
 };
 EXPORT_SYMBOL_GPL(cs42xx8_pm);
-- 
cgit v1.2.3-59-g8ed1b


From 18291410557fb860706d1889505cc15d6e31d790 Mon Sep 17 00:00:00 2001
From: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Date: Sat, 24 Aug 2019 22:26:52 +0200
Subject: ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Allow SSC to be used on platforms described using audio-graph-card
in Device Tree.

Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/233d5461f4448df151755de7b69a0cd3ad310d5c.1566677788.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/Kconfig | 30 ++++++++++++++++++------------
 1 file changed, 18 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 06c1d5ce642c..f118c229ed82 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -12,25 +12,31 @@ if SND_ATMEL_SOC
 config SND_ATMEL_SOC_PDC
 	tristate
 	depends on HAS_DMA
-	default m if SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=m
-	default y if SND_ATMEL_SOC_SSC_PDC=y || (SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=y)
-
-config SND_ATMEL_SOC_SSC_PDC
-	tristate
 
 config SND_ATMEL_SOC_DMA
 	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
-	default m if SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=m
-	default y if SND_ATMEL_SOC_SSC_DMA=y || (SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=y)
-
-config SND_ATMEL_SOC_SSC_DMA
-	tristate
 
 config SND_ATMEL_SOC_SSC
 	tristate
-	default y if SND_ATMEL_SOC_SSC_DMA=y || SND_ATMEL_SOC_SSC_PDC=y
-	default m if SND_ATMEL_SOC_SSC_DMA=m || SND_ATMEL_SOC_SSC_PDC=m
+
+config SND_ATMEL_SOC_SSC_PDC
+	tristate "SoC PCM DAI support for AT91 SSC controller using PDC"
+	depends on ATMEL_SSC
+	select SND_ATMEL_SOC_PDC
+	select SND_ATMEL_SOC_SSC
+	help
+	  Say Y or M if you want to add support for Atmel SSC interface
+	  in PDC mode configured using audio-graph-card in device-tree.
+
+config SND_ATMEL_SOC_SSC_DMA
+	tristate "SoC PCM DAI support for AT91 SSC controller using DMA"
+	depends on ATMEL_SSC
+	select SND_ATMEL_SOC_DMA
+	select SND_ATMEL_SOC_SSC
+	help
+	  Say Y or M if you want to add support for Atmel SSC interface
+	  in DMA mode configured using audio-graph-card in device-tree.
 
 config SND_AT91_SOC_SAM9G20_WM8731
 	tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
-- 
cgit v1.2.3-59-g8ed1b


From db8e3e2091c3ccc4e1dde0f78b3111301cf129e9 Mon Sep 17 00:00:00 2001
From: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Date: Sat, 24 Aug 2019 22:26:53 +0200
Subject: ASoC: atmel_ssc_dai: rework DAI format configuration
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Rework DAI format calculation in preparation for adding more formats
later. As a side-effect this enables all CBM/CBS x CFM/CFS combinations
for supported formats. (Note: the additional modes are not tested.)

Note: this changes FSEDGE to POSITIVE for I2S CBM_CFS mode as the TXSYN
interrupt is not used anyway.

Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Link: https://lore.kernel.org/r/f5949b0326fdcdca072f3ed03f77de9e207631cd.1566677788.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/atmel_ssc_dai.c | 288 +++++++++++-----------------------------
 1 file changed, 81 insertions(+), 207 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 6f89483ac88c..7dc6ec9b8c7a 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -471,7 +471,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	int dir, channels, bits;
 	u32 tfmr, rfmr, tcmr, rcmr;
 	int ret;
-	int fslen, fslen_ext;
+	int fslen, fslen_ext, fs_osync, fs_edge;
 	u32 cmr_div;
 	u32 tcmr_period;
 	u32 rcmr_period;
@@ -558,233 +558,107 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	/*
 	 * Compute SSC register settings.
 	 */
-	switch (ssc_p->daifmt
-		& (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
 
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+	fslen_ext = (bits - 1) / 16;
+	fslen = (bits - 1) % 16;
+
+	switch (ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+
+	case SND_SOC_DAIFMT_I2S:
+		fs_osync = SSC_FSOS_NEGATIVE;
+		fs_edge = SSC_START_FALLING_RF;
+
+		rcmr =	  SSC_BF(RCMR_STTDLY, 1);
+		tcmr =	  SSC_BF(TCMR_STTDLY, 1);
+
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
 		/*
-		 * I2S format, SSC provides BCLK and LRC clocks.
+		 * DSP/PCM Mode A format
 		 *
-		 * The SSC transmit and receive clocks are generated
-		 * from the MCK divider, and the BCLK signal
-		 * is output on the SSC TK line.
+		 * Data is transferred on first BCLK after LRC pulse rising
+		 * edge.If stereo, the right channel data is contiguous with
+		 * the left channel data.
 		 */
+		fs_osync = SSC_FSOS_POSITIVE;
+		fs_edge = SSC_START_RISING_RF;
+		fslen = fslen_ext = 0;
 
-		if (bits > 16 && !ssc->pdata->has_fslen_ext) {
-			dev_err(dai->dev,
-				"sample size %d is too large for SSC device\n",
-				bits);
-			return -EINVAL;
-		}
+		rcmr =	  SSC_BF(RCMR_STTDLY, 1);
+		tcmr =	  SSC_BF(TCMR_STTDLY, 1);
 
-		fslen_ext = (bits - 1) / 16;
-		fslen = (bits - 1) % 16;
-
-		rcmr =	  SSC_BF(RCMR_PERIOD, rcmr_period)
-			| SSC_BF(RCMR_STTDLY, START_DELAY)
-			| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
-			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
-
-		rfmr =    SSC_BF(RFMR_FSLEN_EXT, fslen_ext)
-			| SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
-			| SSC_BF(RFMR_FSLEN, fslen)
-			| SSC_BF(RFMR_DATNB, (channels - 1))
-			| SSC_BIT(RFMR_MSBF)
-			| SSC_BF(RFMR_LOOP, 0)
-			| SSC_BF(RFMR_DATLEN, (bits - 1));
-
-		tcmr =	  SSC_BF(TCMR_PERIOD, tcmr_period)
-			| SSC_BF(TCMR_STTDLY, START_DELAY)
-			| SSC_BF(TCMR_START, SSC_START_FALLING_RF)
-			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
-			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
-			| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
-
-		tfmr =    SSC_BF(TFMR_FSLEN_EXT, fslen_ext)
-			| SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(TFMR_FSDEN, 0)
-			| SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
-			| SSC_BF(TFMR_FSLEN, fslen)
-			| SSC_BF(TFMR_DATNB, (channels - 1))
-			| SSC_BIT(TFMR_MSBF)
-			| SSC_BF(TFMR_DATDEF, 0)
-			| SSC_BF(TFMR_DATLEN, (bits - 1));
 		break;
 
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
-		/* I2S format, CODEC supplies BCLK and LRC clocks. */
-		rcmr =	  SSC_BF(RCMR_PERIOD, 0)
-			| SSC_BF(RCMR_STTDLY, START_DELAY)
-			| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
-			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
-					   SSC_CKS_PIN : SSC_CKS_CLOCK);
-
-		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
-			| SSC_BF(RFMR_FSLEN, 0)
-			| SSC_BF(RFMR_DATNB, (channels - 1))
-			| SSC_BIT(RFMR_MSBF)
-			| SSC_BF(RFMR_LOOP, 0)
-			| SSC_BF(RFMR_DATLEN, (bits - 1));
-
-		tcmr =	  SSC_BF(TCMR_PERIOD, 0)
-			| SSC_BF(TCMR_STTDLY, START_DELAY)
-			| SSC_BF(TCMR_START, SSC_START_FALLING_RF)
-			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
-			| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ?
-					   SSC_CKS_CLOCK : SSC_CKS_PIN);
-
-		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(TFMR_FSDEN, 0)
-			| SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
-			| SSC_BF(TFMR_FSLEN, 0)
-			| SSC_BF(TFMR_DATNB, (channels - 1))
-			| SSC_BIT(TFMR_MSBF)
-			| SSC_BF(TFMR_DATDEF, 0)
-			| SSC_BF(TFMR_DATLEN, (bits - 1));
-		break;
+	default:
+		printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
+			ssc_p->daifmt);
+		return -EINVAL;
+	}
 
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFS:
-		/* I2S format, CODEC supplies BCLK, SSC supplies LRCLK. */
-		if (bits > 16 && !ssc->pdata->has_fslen_ext) {
-			dev_err(dai->dev,
-				"sample size %d is too large for SSC device\n",
-				bits);
-			return -EINVAL;
-		}
+	if (!atmel_ssc_cfs(ssc_p)) {
+		fslen = fslen_ext = 0;
+		rcmr_period = tcmr_period = 0;
+		fs_osync = SSC_FSOS_NONE;
+	}
 
-		fslen_ext = (bits - 1) / 16;
-		fslen = (bits - 1) % 16;
-
-		rcmr =	  SSC_BF(RCMR_PERIOD, rcmr_period)
-			| SSC_BF(RCMR_STTDLY, START_DELAY)
-			| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
-			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
-					   SSC_CKS_PIN : SSC_CKS_CLOCK);
-
-		rfmr =    SSC_BF(RFMR_FSLEN_EXT, fslen_ext)
-			| SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
-			| SSC_BF(RFMR_FSLEN, fslen)
-			| SSC_BF(RFMR_DATNB, (channels - 1))
-			| SSC_BIT(RFMR_MSBF)
-			| SSC_BF(RFMR_LOOP, 0)
-			| SSC_BF(RFMR_DATLEN, (bits - 1));
-
-		tcmr =	  SSC_BF(TCMR_PERIOD, tcmr_period)
-			| SSC_BF(TCMR_STTDLY, START_DELAY)
-			| SSC_BF(TCMR_START, SSC_START_FALLING_RF)
-			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
-			| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ?
-					   SSC_CKS_CLOCK : SSC_CKS_PIN);
-
-		tfmr =    SSC_BF(TFMR_FSLEN_EXT, fslen_ext)
-			| SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_NEGATIVE)
-			| SSC_BF(TFMR_FSDEN, 0)
-			| SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
-			| SSC_BF(TFMR_FSLEN, fslen)
-			| SSC_BF(TFMR_DATNB, (channels - 1))
-			| SSC_BIT(TFMR_MSBF)
-			| SSC_BF(TFMR_DATDEF, 0)
-			| SSC_BF(TFMR_DATLEN, (bits - 1));
-		break;
+	rcmr |=	  SSC_BF(RCMR_START, fs_edge);
+	tcmr |=	  SSC_BF(TCMR_START, fs_edge);
 
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+	if (atmel_ssc_cbs(ssc_p)) {
 		/*
-		 * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
+		 * SSC provides BCLK
 		 *
 		 * The SSC transmit and receive clocks are generated from the
 		 * MCK divider, and the BCLK signal is output
 		 * on the SSC TK line.
 		 */
-		rcmr =	  SSC_BF(RCMR_PERIOD, rcmr_period)
-			| SSC_BF(RCMR_STTDLY, 1)
-			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
-			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
-
-		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE)
-			| SSC_BF(RFMR_FSLEN, 0)
-			| SSC_BF(RFMR_DATNB, (channels - 1))
-			| SSC_BIT(RFMR_MSBF)
-			| SSC_BF(RFMR_LOOP, 0)
-			| SSC_BF(RFMR_DATLEN, (bits - 1));
-
-		tcmr =	  SSC_BF(TCMR_PERIOD, tcmr_period)
-			| SSC_BF(TCMR_STTDLY, 1)
-			| SSC_BF(TCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
-			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
-			| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
-
-		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(TFMR_FSDEN, 0)
-			| SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE)
-			| SSC_BF(TFMR_FSLEN, 0)
-			| SSC_BF(TFMR_DATNB, (channels - 1))
-			| SSC_BIT(TFMR_MSBF)
-			| SSC_BF(TFMR_DATDEF, 0)
-			| SSC_BF(TFMR_DATLEN, (bits - 1));
-		break;
+		rcmr |=	  SSC_BF(RCMR_CKS, SSC_CKS_DIV)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE);
 
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
-		/*
-		 * DSP/PCM Mode A format, CODEC supplies BCLK and LRC clocks.
-		 *
-		 * Data is transferred on first BCLK after LRC pulse rising
-		 * edge.If stereo, the right channel data is contiguous with
-		 * the left channel data.
-		 */
-		rcmr =	  SSC_BF(RCMR_PERIOD, 0)
-			| SSC_BF(RCMR_STTDLY, START_DELAY)
-			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
-			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
-					   SSC_CKS_PIN : SSC_CKS_CLOCK);
-
-		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
-			| SSC_BF(RFMR_FSLEN, 0)
-			| SSC_BF(RFMR_DATNB, (channels - 1))
-			| SSC_BIT(RFMR_MSBF)
-			| SSC_BF(RFMR_LOOP, 0)
-			| SSC_BF(RFMR_DATLEN, (bits - 1));
-
-		tcmr =	  SSC_BF(TCMR_PERIOD, 0)
-			| SSC_BF(TCMR_STTDLY, START_DELAY)
-			| SSC_BF(TCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
-			| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
-			| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
-					   SSC_CKS_CLOCK : SSC_CKS_PIN);
-
-		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
-			| SSC_BF(TFMR_FSDEN, 0)
-			| SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
-			| SSC_BF(TFMR_FSLEN, 0)
-			| SSC_BF(TFMR_DATNB, (channels - 1))
-			| SSC_BIT(TFMR_MSBF)
-			| SSC_BF(TFMR_DATDEF, 0)
-			| SSC_BF(TFMR_DATLEN, (bits - 1));
-		break;
+		tcmr |=	  SSC_BF(TCMR_CKS, SSC_CKS_DIV)
+			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS);
+	} else {
+		rcmr |=	  SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+					SSC_CKS_PIN : SSC_CKS_CLOCK)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE);
 
-	default:
-		printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
-			ssc_p->daifmt);
+		tcmr |=	  SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ?
+					SSC_CKS_CLOCK : SSC_CKS_PIN)
+			| SSC_BF(TCMR_CKO, SSC_CKO_NONE);
+	}
+
+	rcmr |=	  SSC_BF(RCMR_PERIOD, rcmr_period)
+		| SSC_BF(RCMR_CKI, SSC_CKI_RISING);
+
+	tcmr |=   SSC_BF(TCMR_PERIOD, tcmr_period)
+		| SSC_BF(TCMR_CKI, SSC_CKI_FALLING);
+
+	rfmr =    SSC_BF(RFMR_FSLEN_EXT, fslen_ext)
+		| SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+		| SSC_BF(RFMR_FSOS, fs_osync)
+		| SSC_BF(RFMR_FSLEN, fslen)
+		| SSC_BF(RFMR_DATNB, (channels - 1))
+		| SSC_BIT(RFMR_MSBF)
+		| SSC_BF(RFMR_LOOP, 0)
+		| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+	tfmr =    SSC_BF(TFMR_FSLEN_EXT, fslen_ext)
+		| SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+		| SSC_BF(TFMR_FSDEN, 0)
+		| SSC_BF(TFMR_FSOS, fs_osync)
+		| SSC_BF(TFMR_FSLEN, fslen)
+		| SSC_BF(TFMR_DATNB, (channels - 1))
+		| SSC_BIT(TFMR_MSBF)
+		| SSC_BF(TFMR_DATDEF, 0)
+		| SSC_BF(TFMR_DATLEN, (bits - 1));
+
+	if (fslen_ext && !ssc->pdata->has_fslen_ext) {
+		dev_err(dai->dev, "sample size %d is too large for SSC device\n",
+			bits);
 		return -EINVAL;
 	}
+
 	pr_debug("atmel_ssc_hw_params: "
 			"RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
 			rcmr, rfmr, tcmr, tfmr);
-- 
cgit v1.2.3-59-g8ed1b


From 376142b7bbc1edef2900ddc300a006f8c056b0d5 Mon Sep 17 00:00:00 2001
From: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Date: Sat, 24 Aug 2019 22:26:54 +0200
Subject: ASoC: atmel_ssc_dai: implement left-justified data mode
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Enable support for left-justified data mode for SSC-codec link.

Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/44fa6b700421e80778f20ff9ead2b148cf6d2e92.1566677788.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/atmel/atmel_ssc_dai.c | 9 +++++++++
 1 file changed, 9 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 7dc6ec9b8c7a..48e9eef34c0f 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -564,6 +564,15 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 
 	switch (ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 
+	case SND_SOC_DAIFMT_LEFT_J:
+		fs_osync = SSC_FSOS_POSITIVE;
+		fs_edge = SSC_START_RISING_RF;
+
+		rcmr =	  SSC_BF(RCMR_STTDLY, 0);
+		tcmr =	  SSC_BF(TCMR_STTDLY, 0);
+
+		break;
+
 	case SND_SOC_DAIFMT_I2S:
 		fs_osync = SSC_FSOS_NEGATIVE;
 		fs_edge = SSC_START_FALLING_RF;
-- 
cgit v1.2.3-59-g8ed1b


From eb7505d52a2f8b0cfc3fd7146d8cb2dab5a73f0d Mon Sep 17 00:00:00 2001
From: Jussi Laako <jussi@sonarnerd.net>
Date: Wed, 28 Aug 2019 00:08:46 +0300
Subject: ALSA: usb-audio: DSD auto-detection for Playback Designs

Add DSD support auto-detection for newer Playback Designs devices. Older
device generations have a different USB interface implementation.

Keep the auto-detection VID whitelist sorted.

Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 5c0fa5cf1987..08ec3dac5dfe 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1566,7 +1566,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	struct usb_interface *iface;
 
 	/* Playback Designs */
-	if (USB_ID_VENDOR(chip->usb_id) == 0x23ba) {
+	if (USB_ID_VENDOR(chip->usb_id) == 0x23ba &&
+	    USB_ID_PRODUCT(chip->usb_id) < 0x0110) {
 		switch (fp->altsetting) {
 		case 1:
 			fp->dsd_dop = true;
@@ -1654,8 +1655,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	 * from XMOS/Thesycon
 	 */
 	switch (USB_ID_VENDOR(chip->usb_id)) {
-	case 0x20b1:  /* XMOS based devices */
 	case 0x152a:  /* Thesycon devices */
+	case 0x20b1:  /* XMOS based devices */
+	case 0x23ba:  /* Playback Designs */
 	case 0x25ce:  /* Mytek devices */
 	case 0x2ab6:  /* T+A devices */
 		if (fp->dsd_raw)
-- 
cgit v1.2.3-59-g8ed1b


From 0067e154b11e236d62a7a8205f321b097c21a35b Mon Sep 17 00:00:00 2001
From: Jussi Laako <jussi@sonarnerd.net>
Date: Wed, 28 Aug 2019 00:08:47 +0300
Subject: ALSA: usb-audio: Update DSD support quirks for Oppo and Rotel

Oppo has issued firmware updates that change alt setting used for DSD
support. However, these devices seem to support auto-detection, so
support is moved from explicit whitelisting to auto-detection.

Also Rotel devices have USB interfaces that support DSD with
auto-detection.

Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 08ec3dac5dfe..25faf2d3c639 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1584,9 +1584,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	/* XMOS based USB DACs */
 	switch (chip->usb_id) {
 	case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */
-	case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */
-	case USB_ID(0x22d9, 0x0436): /* OPPO Sonica */
-	case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */
 	case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */
 	case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */
 		if (fp->altsetting == 2)
@@ -1600,7 +1597,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */
 	case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */
 	case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */
-	case USB_ID(0x22d9, 0x0426): /* OPPO HA-2 */
 	case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */
 	case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */
 	case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */
@@ -1657,8 +1653,10 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	switch (USB_ID_VENDOR(chip->usb_id)) {
 	case 0x152a:  /* Thesycon devices */
 	case 0x20b1:  /* XMOS based devices */
+	case 0x22d9:  /* Oppo */
 	case 0x23ba:  /* Playback Designs */
 	case 0x25ce:  /* Mytek devices */
+	case 0x278b:  /* Rotel? */
 	case 0x2ab6:  /* T+A devices */
 		if (fp->dsd_raw)
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
-- 
cgit v1.2.3-59-g8ed1b


From 8e58c521bcb823bfd6e81f4a85b76afdce098b1d Mon Sep 17 00:00:00 2001
From: Jiaxin Yu <jiaxin.yu@mediatek.com>
Date: Sat, 24 Aug 2019 12:37:24 +0800
Subject: ASoC: mediatek: mt8183: tdm hw support tdm out and 8ch i2s out

This patch refined tdm driver code, and allow tdm hw to support two
configurations in machine driver to output tdm signal or i2s signal.

Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1566621445-26989-3-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-dai-tdm.c         | 175 +++++++++++++++++----
 .../mt8183/mt8183-mt6358-ts3a227-max98357.c        |   3 +
 2 files changed, 144 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c b/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c
index 8983d54a9b67..d34cabdbf889 100644
--- a/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c
+++ b/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c
@@ -15,13 +15,30 @@
 struct mtk_afe_tdm_priv {
 	int bck_id;
 	int bck_rate;
-
+	int tdm_out_mode;
+	int bck_invert;
+	int lck_invert;
 	int mclk_id;
 	int mclk_multiple; /* according to sample rate */
 	int mclk_rate;
 	int mclk_apll;
 };
 
+enum {
+	TDM_OUT_I2S = 0,
+	TDM_OUT_TDM = 1,
+};
+
+enum {
+	TDM_BCK_NON_INV = 0,
+	TDM_BCK_INV = 1,
+};
+
+enum {
+	TDM_LCK_NON_INV = 0,
+	TDM_LCK_INV = 1,
+};
+
 enum {
 	TDM_WLEN_16_BIT = 1,
 	TDM_WLEN_32_BIT = 2,
@@ -93,6 +110,25 @@ static unsigned int get_tdm_ch(unsigned int ch)
 	}
 }
 
+static unsigned int get_tdm_ch_fixup(unsigned int channels)
+{
+	if (channels > 4)
+		return 8;
+	else if (channels > 2)
+		return 4;
+	else
+		return 2;
+}
+
+static unsigned int get_tdm_ch_per_sdata(unsigned int mode,
+					 unsigned int channels)
+{
+	if (mode == TDM_OUT_TDM)
+		return get_tdm_ch_fixup(channels);
+	else
+		return 2;
+}
+
 /* interconnection */
 enum {
 	HDMI_CONN_CH0 = 0,
@@ -433,8 +469,11 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
 	struct mt8183_afe_private *afe_priv = afe->platform_priv;
 	int tdm_id = dai->id;
 	struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[tdm_id];
+	unsigned int tdm_out_mode = tdm_priv->tdm_out_mode;
 	unsigned int rate = params_rate(params);
 	unsigned int channels = params_channels(params);
+	unsigned int out_channels_per_sdata =
+		get_tdm_ch_per_sdata(tdm_out_mode, channels);
 	snd_pcm_format_t format = params_format(params);
 	unsigned int tdm_con = 0;
 
@@ -448,7 +487,7 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
 
 	/* calculate bck */
 	tdm_priv->bck_rate = rate *
-			     channels *
+			     out_channels_per_sdata *
 			     snd_pcm_format_physical_width(format);
 
 	if (tdm_priv->bck_rate > tdm_priv->mclk_rate)
@@ -461,50 +500,70 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
 		 __func__,
 		 tdm_id, rate, channels, format,
 		 tdm_priv->mclk_rate, tdm_priv->bck_rate);
+	dev_info(afe->dev, "%s(), out_channels_per_sdata = %d\n",
+		 __func__, out_channels_per_sdata);
 
 	/* set tdm */
-	tdm_con = 1 << BCK_INVERSE_SFT;
-	tdm_con |= 1 << LRCK_INVERSE_SFT;
-	tdm_con |= 1 << DELAY_DATA_SFT;
+	if (tdm_priv->bck_invert)
+		tdm_con |= 1 << BCK_INVERSE_SFT;
+
+	if (tdm_priv->lck_invert)
+		tdm_con |= 1 << LRCK_INVERSE_SFT;
+
+	if (tdm_priv->tdm_out_mode == TDM_OUT_I2S) {
+		tdm_con |= 1 << DELAY_DATA_SFT;
+		tdm_con |= get_tdm_lrck_width(format) << LRCK_TDM_WIDTH_SFT;
+	} else if (tdm_priv->tdm_out_mode == TDM_OUT_TDM) {
+		tdm_con |= 0 << DELAY_DATA_SFT;
+		tdm_con |= 0 << LRCK_TDM_WIDTH_SFT;
+	}
+
 	tdm_con |= 1 << LEFT_ALIGN_SFT;
 	tdm_con |= get_tdm_wlen(format) << WLEN_SFT;
-	tdm_con |= get_tdm_ch(channels) << CHANNEL_NUM_SFT;
+	tdm_con |= get_tdm_ch(out_channels_per_sdata) << CHANNEL_NUM_SFT;
 	tdm_con |= get_tdm_channel_bck(format) << CHANNEL_BCK_CYCLES_SFT;
-	tdm_con |= get_tdm_lrck_width(format) << LRCK_TDM_WIDTH_SFT;
 	regmap_write(afe->regmap, AFE_TDM_CON1, tdm_con);
 
-	switch (channels) {
-	case 1:
-	case 2:
+	if (out_channels_per_sdata == 2) {
+		switch (channels) {
+		case 1:
+		case 2:
+			tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
+			tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT1_SFT;
+			tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT2_SFT;
+			tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT3_SFT;
+			break;
+		case 3:
+		case 4:
+			tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
+			tdm_con |= TDM_CH_START_O32_O33 << ST_CH_PAIR_SOUT1_SFT;
+			tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT2_SFT;
+			tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT3_SFT;
+			break;
+		case 5:
+		case 6:
+			tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
+			tdm_con |= TDM_CH_START_O32_O33 << ST_CH_PAIR_SOUT1_SFT;
+			tdm_con |= TDM_CH_START_O34_O35 << ST_CH_PAIR_SOUT2_SFT;
+			tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT3_SFT;
+			break;
+		case 7:
+		case 8:
+			tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
+			tdm_con |= TDM_CH_START_O32_O33 << ST_CH_PAIR_SOUT1_SFT;
+			tdm_con |= TDM_CH_START_O34_O35 << ST_CH_PAIR_SOUT2_SFT;
+			tdm_con |= TDM_CH_START_O36_O37 << ST_CH_PAIR_SOUT3_SFT;
+			break;
+		default:
+			tdm_con = 0;
+		}
+	} else {
 		tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
 		tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT1_SFT;
 		tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT2_SFT;
 		tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT3_SFT;
-		break;
-	case 3:
-	case 4:
-		tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
-		tdm_con |= TDM_CH_START_O32_O33 << ST_CH_PAIR_SOUT1_SFT;
-		tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT2_SFT;
-		tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT3_SFT;
-		break;
-	case 5:
-	case 6:
-		tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
-		tdm_con |= TDM_CH_START_O32_O33 << ST_CH_PAIR_SOUT1_SFT;
-		tdm_con |= TDM_CH_START_O34_O35 << ST_CH_PAIR_SOUT2_SFT;
-		tdm_con |= TDM_CH_ZERO << ST_CH_PAIR_SOUT3_SFT;
-		break;
-	case 7:
-	case 8:
-		tdm_con = TDM_CH_START_O30_O31 << ST_CH_PAIR_SOUT0_SFT;
-		tdm_con |= TDM_CH_START_O32_O33 << ST_CH_PAIR_SOUT1_SFT;
-		tdm_con |= TDM_CH_START_O34_O35 << ST_CH_PAIR_SOUT2_SFT;
-		tdm_con |= TDM_CH_START_O36_O37 << ST_CH_PAIR_SOUT3_SFT;
-		break;
-	default:
-		tdm_con = 0;
 	}
+
 	regmap_write(afe->regmap, AFE_TDM_CON2, tdm_con);
 
 	regmap_update_bits(afe->regmap, AFE_HDMI_OUT_CON0,
@@ -573,10 +632,58 @@ static int mtk_dai_tdm_set_sysclk(struct snd_soc_dai *dai,
 	return mtk_dai_tdm_cal_mclk(afe, tdm_priv, freq);
 }
 
+static int mtk_dai_tdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct mtk_base_afe *afe = dev_get_drvdata(dai->dev);
+	struct mt8183_afe_private *afe_priv = afe->platform_priv;
+	struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id];
+
+	if (!tdm_priv) {
+		dev_warn(afe->dev, "%s(), tdm_priv == NULL", __func__);
+		return -EINVAL;
+	}
+
+	/* DAI mode*/
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		tdm_priv->tdm_out_mode = TDM_OUT_I2S;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		tdm_priv->tdm_out_mode = TDM_OUT_TDM;
+		break;
+	default:
+		tdm_priv->tdm_out_mode = TDM_OUT_I2S;
+	}
+
+	/* DAI clock inversion*/
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		tdm_priv->bck_invert = TDM_BCK_NON_INV;
+		tdm_priv->lck_invert = TDM_LCK_NON_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		tdm_priv->bck_invert = TDM_BCK_NON_INV;
+		tdm_priv->lck_invert = TDM_LCK_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		tdm_priv->bck_invert = TDM_BCK_INV;
+		tdm_priv->lck_invert = TDM_LCK_NON_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+	default:
+		tdm_priv->bck_invert = TDM_BCK_INV;
+		tdm_priv->lck_invert = TDM_LCK_INV;
+		break;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dai_ops mtk_dai_tdm_ops = {
 	.hw_params = mtk_dai_tdm_hw_params,
 	.trigger = mtk_dai_tdm_trigger,
 	.set_sysclk = mtk_dai_tdm_set_sysclk,
+	.set_fmt = mtk_dai_tdm_set_fmt,
 };
 
 /* dai driver */
diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 951706fa75fc..560fcc5923a4 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -380,6 +380,9 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = {
 	{
 		.name = "TDM",
 		.no_pcm = 1,
+		.dai_fmt = SND_SOC_DAIFMT_I2S |
+			   SND_SOC_DAIFMT_IB_IF |
+			   SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_playback = 1,
 		.ignore_suspend = 1,
 		.be_hw_params_fixup = mt8183_i2s_hw_params_fixup,
-- 
cgit v1.2.3-59-g8ed1b


From 2c650fd5a53beace2e7507f478562f0dd1885b89 Mon Sep 17 00:00:00 2001
From: Jiaxin Yu <jiaxin.yu@mediatek.com>
Date: Sat, 24 Aug 2019 12:37:25 +0800
Subject: ASoC: mediatek: mt8183: fix tdm out data is valid on rising edge

This patch correct tdm out bck inverse register to AUDIO_TOP_CON3[3].

Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Link: https://lore.kernel.org/r/1566621445-26989-4-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 4 +++-
 sound/soc/mediatek/mt8183/mt8183-reg.h     | 8 +++++---
 2 files changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c b/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c
index d34cabdbf889..0d69cf440407 100644
--- a/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c
+++ b/sound/soc/mediatek/mt8183/mt8183-dai-tdm.c
@@ -505,7 +505,9 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream,
 
 	/* set tdm */
 	if (tdm_priv->bck_invert)
-		tdm_con |= 1 << BCK_INVERSE_SFT;
+		regmap_update_bits(afe->regmap, AUDIO_TOP_CON3,
+				   BCK_INVERSE_MASK_SFT,
+				   0x1 << BCK_INVERSE_SFT);
 
 	if (tdm_priv->lck_invert)
 		tdm_con |= 1 << LRCK_INVERSE_SFT;
diff --git a/sound/soc/mediatek/mt8183/mt8183-reg.h b/sound/soc/mediatek/mt8183/mt8183-reg.h
index e0482f2826da..e544a09e1913 100644
--- a/sound/soc/mediatek/mt8183/mt8183-reg.h
+++ b/sound/soc/mediatek/mt8183/mt8183-reg.h
@@ -413,6 +413,11 @@
 #define AFE_MAX_REGISTER AFE_GENERAL2_ASRC_2CH_CON13
 #define AFE_IRQ_STATUS_BITS 0x1fff
 
+/* AUDIO_TOP_CON3 */
+#define BCK_INVERSE_SFT                              3
+#define BCK_INVERSE_MASK                             0x1
+#define BCK_INVERSE_MASK_SFT                         (0x1 << 3)
+
 /* AFE_DAC_CON0 */
 #define AWB2_ON_SFT                                   29
 #define AWB2_ON_MASK                                  0x1
@@ -1596,9 +1601,6 @@
 #define TDM_EN_SFT                                    0
 #define TDM_EN_MASK                                   0x1
 #define TDM_EN_MASK_SFT                               (0x1 << 0)
-#define BCK_INVERSE_SFT                               1
-#define BCK_INVERSE_MASK                              0x1
-#define BCK_INVERSE_MASK_SFT                          (0x1 << 1)
 #define LRCK_INVERSE_SFT                              2
 #define LRCK_INVERSE_MASK                             0x1
 #define LRCK_INVERSE_MASK_SFT                         (0x1 << 2)
-- 
cgit v1.2.3-59-g8ed1b


From 4cc4531c310e592cf624148ae59c64f930f12e39 Mon Sep 17 00:00:00 2001
From: Vidyakumar Athota <vathota@codeaurora.org>
Date: Thu, 22 Aug 2019 10:56:50 +0100
Subject: ALSA: pcm: add support for 352.8KHz and 384KHz sample rate

Most of the modern codecs supports 352.8KHz and 384KHz sample rates.
Currenlty HW params fails to set 352.8Kz and 384KHz sample rate
as these are not in known rates list.
Add these new rates to known list to allow them.

This patch also adds defines in pcm.h so that drivers can use it.

Signed-off-by: Vidyakumar Athota <vathota@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20190822095653.7200-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/pcm.h     | 5 +++++
 sound/core/pcm_native.c | 2 +-
 2 files changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 1e9bb1c91770..bbe6eb1ff5d2 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -117,6 +117,8 @@ struct snd_pcm_ops {
 #define SNDRV_PCM_RATE_96000		(1<<10)		/* 96000Hz */
 #define SNDRV_PCM_RATE_176400		(1<<11)		/* 176400Hz */
 #define SNDRV_PCM_RATE_192000		(1<<12)		/* 192000Hz */
+#define SNDRV_PCM_RATE_352800		(1<<13)		/* 352800Hz */
+#define SNDRV_PCM_RATE_384000		(1<<14)		/* 384000Hz */
 
 #define SNDRV_PCM_RATE_CONTINUOUS	(1<<30)		/* continuous range */
 #define SNDRV_PCM_RATE_KNOT		(1<<31)		/* supports more non-continuos rates */
@@ -129,6 +131,9 @@ struct snd_pcm_ops {
 					 SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000)
 #define SNDRV_PCM_RATE_8000_192000	(SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\
 					 SNDRV_PCM_RATE_192000)
+#define SNDRV_PCM_RATE_8000_384000	(SNDRV_PCM_RATE_8000_192000|\
+					 SNDRV_PCM_RATE_352800|\
+					 SNDRV_PCM_RATE_384000)
 #define _SNDRV_PCM_FMTBIT(fmt)		(1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt)
 #define SNDRV_PCM_FMTBIT_S8		_SNDRV_PCM_FMTBIT(S8)
 #define SNDRV_PCM_FMTBIT_U8		_SNDRV_PCM_FMTBIT(U8)
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 860543a4c840..34390be3fb0f 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -2168,7 +2168,7 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params,
 
 static const unsigned int rates[] = {
 	5512, 8000, 11025, 16000, 22050, 32000, 44100,
-	48000, 64000, 88200, 96000, 176400, 192000
+	48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000
 };
 
 const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = {
-- 
cgit v1.2.3-59-g8ed1b


From a8a652bfac7f4a60dfbe07a8c95c65506601b2e1 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 22 Aug 2019 10:56:51 +0100
Subject: ASoC: wcd9335: Fix primary interpolator max rate

On this codec SLIMBus RX path supports 384000 rate on primary interpolator.
Add this missing rate as supported rate.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20190822095653.7200-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd9335.c | 20 ++++++++++++--------
 1 file changed, 12 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index 956602788d0e..03f8a94bba2f 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -2071,9 +2071,10 @@ static struct snd_soc_dai_driver wcd9335_slim_dais[] = {
 		.id = AIF1_PB,
 		.playback = {
 			.stream_name = "AIF1 Playback",
-			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK,
+			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK |
+				 SNDRV_PCM_RATE_384000,
 			.formats = WCD9335_FORMATS_S16_S24_LE,
-			.rate_max = 192000,
+			.rate_max = 384000,
 			.rate_min = 8000,
 			.channels_min = 1,
 			.channels_max = 2,
@@ -2099,10 +2100,11 @@ static struct snd_soc_dai_driver wcd9335_slim_dais[] = {
 		.id = AIF2_PB,
 		.playback = {
 			.stream_name = "AIF2 Playback",
-			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK,
+			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK |
+				 SNDRV_PCM_RATE_384000,
 			.formats = WCD9335_FORMATS_S16_S24_LE,
 			.rate_min = 8000,
-			.rate_max = 192000,
+			.rate_max = 384000,
 			.channels_min = 1,
 			.channels_max = 2,
 		},
@@ -2127,10 +2129,11 @@ static struct snd_soc_dai_driver wcd9335_slim_dais[] = {
 		.id = AIF3_PB,
 		.playback = {
 			.stream_name = "AIF3 Playback",
-			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK,
+			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK |
+				 SNDRV_PCM_RATE_384000,
 			.formats = WCD9335_FORMATS_S16_S24_LE,
 			.rate_min = 8000,
-			.rate_max = 192000,
+			.rate_max = 384000,
 			.channels_min = 1,
 			.channels_max = 2,
 		},
@@ -2155,10 +2158,11 @@ static struct snd_soc_dai_driver wcd9335_slim_dais[] = {
 		.id = AIF4_PB,
 		.playback = {
 			.stream_name = "AIF4 Playback",
-			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK,
+			.rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK |
+				 SNDRV_PCM_RATE_384000,
 			.formats = WCD9335_FORMATS_S16_S24_LE,
 			.rate_min = 8000,
-			.rate_max = 192000,
+			.rate_max = 384000,
 			.channels_min = 1,
 			.channels_max = 2,
 		},
-- 
cgit v1.2.3-59-g8ed1b


From 6e26d193d15029a65e0904ad85d6d8634817c3a6 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 29 Aug 2019 23:37:19 +0900
Subject: ALSA: dice: fix error path for initialization of stream structures

When introducing AMDTP domain to ALSA dice driver, error path does not
handle error correctly. This commit fixes the bug.

Fixes: e9f21129b8d8 ("ALSA: dice: support AMDTP domain")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/dice/dice-stream.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index af8c5a2c28f3..f6a8627ae5a2 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -570,7 +570,7 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice)
 				destroy_stream(dice, AMDTP_OUT_STREAM, i);
 			for (i = 0; i < MAX_STREAMS; i++)
 				destroy_stream(dice, AMDTP_IN_STREAM, i);
-			break;
+			goto end;
 		}
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From f155e79ab516c9ed2adb2f337fb4ff58b7474428 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 29 Aug 2019 23:38:08 +0900
Subject: ALSA: firewire-lib: fix isoc cycle count to which rx packet is
 scheduled

When introducing the list of packet descriptor, for rx packet, the
calculation of scheduled isoc cycle is omitted. This commit fixes the
bug.

Fixes: f4f6ae7b7c1f ("ALSA: firewire-lib: use packet descriptor for IT context")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 1a92855c7647..88270257e896 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -732,7 +732,7 @@ static void generate_ideal_pkt_descs(struct amdtp_stream *s,
 		struct pkt_desc *desc = descs + i;
 		unsigned int index = (s->packet_index + i) % QUEUE_LENGTH;
 
-		desc->cycle = compute_cycle_count(*ctx_header);
+		desc->cycle = compute_it_cycle(*ctx_header);
 		desc->syt = calculate_syt(s, desc->cycle);
 		desc->data_blocks = calculate_data_blocks(s, desc->syt);
 
-- 
cgit v1.2.3-59-g8ed1b


From 696d05225cebffd172008d212657be90e823eac0 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Wed, 28 Aug 2019 13:20:17 -0400
Subject: ASoC: fsl_ssi: Fix clock control issue in master mode

The test case is
arecord -Dhw:0 -d 10 -f S16_LE -r 48000 -c 2 temp.wav &
aplay -Dhw:0 -d 30 -f S16_LE -r 48000 -c 2 test.wav

There will be error after end of arecord:
aplay: pcm_write:2051: write error: Input/output error

Capture and Playback work in parallel in master mode, one
substream stops, the other substream is impacted, the
reason is that clock is disabled wrongly.

The clock's reference count is not increased when second
substream starts, the hw_param() function returns in the
beginning because first substream is enabled, then in end
of first substream, the hw_free() disables the clock.

This patch is to move the clock enablement to the place
before checking of the device enablement in hw_param().

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1567012817-12625-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_ssi.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index fa862af25c1a..085855f9b08d 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	u32 wl = SSI_SxCCR_WL(sample_size);
 	int ret;
 
-	/*
-	 * SSI is properly configured if it is enabled and running in
-	 * the synchronous mode; Note that AC97 mode is an exception
-	 * that should set separate configurations for STCCR and SRCCR
-	 * despite running in the synchronous mode.
-	 */
-	if (ssi->streams && ssi->synchronous)
-		return 0;
-
 	if (fsl_ssi_is_i2s_master(ssi)) {
 		ret = fsl_ssi_set_bclk(substream, dai, hw_params);
 		if (ret)
@@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
+	/*
+	 * SSI is properly configured if it is enabled and running in
+	 * the synchronous mode; Note that AC97 mode is an exception
+	 * that should set separate configurations for STCCR and SRCCR
+	 * despite running in the synchronous mode.
+	 */
+	if (ssi->streams && ssi->synchronous)
+		return 0;
+
 	if (!fsl_ssi_is_ac97(ssi)) {
 		/*
 		 * Keep the ssi->i2s_net intact while having a local variable
-- 
cgit v1.2.3-59-g8ed1b


From cc9bbb6cde6c321706f1eff79d077b4a91cd5d12 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Wed, 28 Aug 2019 11:51:02 +0200
Subject: ASoC: Intel: Baytrail: Fix implicit fallthrough warning

Append fallthrough statement to fix warning reported during compilation.

Fixes: b80d19c166c4 ("ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190828095102.15737-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/baytrail/sst-baytrail-pcm.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
index 9cbc982d46a9..54f2ee3010ee 100644
--- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c
+++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
@@ -193,6 +193,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		pdata->restore_stream = false;
+		/* fallthrough */
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		sst_byt_stream_pause(byt, pcm_data->stream);
 		break;
-- 
cgit v1.2.3-59-g8ed1b


From 6af86bdb8ad41f4cf1292d3b10857dc322758328 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Fri, 30 Aug 2019 10:14:56 +0900
Subject: ALSA: firewire-motu: add support for MOTU 4pre

MOTU 4pre was launched in 2012 by MOTU, Inc. This commit allows userspace
applications can transmit and receive PCM frames and MIDI messages for
this model via ALSA PCM interface and RawMidi/Sequencer interfaces.

The device supports MOTU protocol version 3. Unlike the other devices, the
device is simply designed. The size of data block is fixed to 10 quadlets
during available sampling rates (44.1 - 96.0 kHz). Each data block
includes 1 source packet header, 2 data chunks for messages, 8 data chunks
for PCM samples and 2 data chunks for padding to quadlet alignment. The
device has no MIDI, optical, BNC and AES/EBU interfaces.

Like support for the other MOTU devices, the quality of playback sound
is not enough good with periodical noise yet.

$ python2 crpp < ~/git/am-config-rom/motu/motu-4pre.img
               ROM header and bus information block
               -----------------------------------------------------------------
400  041078cc  bus_info_length 4, crc_length 16, crc 30924
404  31333934  bus_name "1394"
408  20ff7000  irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256)
40c  0001f200  company_id 0001f2     |
410  000a41c5  device_id 00000a41c5  | EUI-64 0001f200000a41c5

               root directory
               -----------------------------------------------------------------
414  0004ef04  directory_length 4, crc 61188
418  030001f2  vendor
41c  0c0083c0  node capabilities per IEEE 1394
420  d1000002  --> unit directory at 428
424  8d000005  --> eui-64 leaf at 438

               unit directory at 428
               -----------------------------------------------------------------
428  0003ceda  directory_length 3, crc 52954
42c  120001f2  specifier id
430  13000045  version
434  17103800  model

               eui-64 leaf at 438
               -----------------------------------------------------------------
438  0002d248  leaf_length 2, crc 53832
43c  0001f200  company_id 0001f2     |
440  000a41c5  device_id 00000a41c5  | EUI-64 0001f200000a41c5

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/motu/motu.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound')

diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c
index 03cda2166ea3..72908b4de77c 100644
--- a/sound/firewire/motu/motu.c
+++ b/sound/firewire/motu/motu.c
@@ -247,6 +247,17 @@ static const struct snd_motu_spec motu_audio_express = {
 	.analog_out_ports = 4,
 };
 
+static const struct snd_motu_spec motu_4pre = {
+	.name = "4pre",
+	.protocol = &snd_motu_protocol_v3,
+	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
+		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
+		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
+		 SND_MOTU_SPEC_RX_SEPARETED_MAIN,
+	.analog_in_ports = 2,
+	.analog_out_ports = 2,
+};
+
 #define SND_MOTU_DEV_ENTRY(model, data)			\
 {							\
 	.match_flags	= IEEE1394_MATCH_VENDOR_ID |	\
@@ -265,6 +276,7 @@ static const struct ieee1394_device_id motu_id_table[] = {
 	SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3),	/* FireWire only. */
 	SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3),	/* Hybrid. */
 	SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express),
+	SND_MOTU_DEV_ENTRY(0x000045, &motu_4pre),
 	{ }
 };
 MODULE_DEVICE_TABLE(ieee1394, motu_id_table);
-- 
cgit v1.2.3-59-g8ed1b


From a2dc6f82fd86fa165222f6062e2478fd122f9f1c Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 30 Aug 2019 13:38:39 +0300
Subject: ASoC: ti: davinci-i2s: Move the XSYNCERR workaround to .prepare
 callback

Currently the driver uses snd_soc_rtdcom_lookup() in it's mcbsp_start
function to try to stop/restart the DMA as the initial XSYNCERR workaround
need to be done before the DMA is armed.

There are couple of things wrong with this:
- the driver crashes with NULL pointer dereference as the
  component->driver->ops is actually NULL
- the driver should not use snd_soc_rtdcom_lookup() in the first place
- Fiddling with DMA is never a good thing

Move the workaround handling to .prepare which is called before the DMA is
armed, so it complies with the requirements.

Reported-by (usage of snd_soc_rtdcom_lookup): Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-i2s.c | 82 +++++++++++++++++++---------------------------
 1 file changed, 34 insertions(+), 48 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 92c1bdc69086..27afdbb9adf3 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -187,57 +187,9 @@ static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback)
 static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev,
 		struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 	u32 spcr;
 	u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST;
-	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
-	if (spcr & mask) {
-		/* start off disabled */
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
-				spcr & ~mask);
-		toggle_clock(dev, playback);
-	}
-	if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM |
-			DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) {
-		/* Start the sample generator */
-		spcr |= DAVINCI_MCBSP_SPCR_GRST;
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
-	}
-
-	if (playback) {
-		/* Stop the DMA to avoid data loss */
-		/* while the transmitter is out of reset to handle XSYNCERR */
-		if (component->driver->ops->trigger) {
-			int ret = component->driver->ops->trigger(substream,
-				SNDRV_PCM_TRIGGER_STOP);
-			if (ret < 0)
-				printk(KERN_DEBUG "Playback DMA stop failed\n");
-		}
-
-		/* Enable the transmitter */
-		spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
-		spcr |= DAVINCI_MCBSP_SPCR_XRST;
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
-
-		/* wait for any unexpected frame sync error to occur */
-		udelay(100);
-
-		/* Disable the transmitter to clear any outstanding XSYNCERR */
-		spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
-		spcr &= ~DAVINCI_MCBSP_SPCR_XRST;
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
-		toggle_clock(dev, playback);
-
-		/* Restart the DMA */
-		if (component->driver->ops->trigger) {
-			int ret = component->driver->ops->trigger(substream,
-				SNDRV_PCM_TRIGGER_START);
-			if (ret < 0)
-				printk(KERN_DEBUG "Playback DMA start failed\n");
-		}
-	}
 
 	/* Enable transmitter or receiver */
 	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
@@ -575,7 +527,41 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream,
 {
 	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
 	int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	u32 spcr;
+	u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST;
+
 	davinci_mcbsp_stop(dev, playback);
+
+	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+	if (spcr & mask) {
+		/* start off disabled */
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
+					spcr & ~mask);
+		toggle_clock(dev, playback);
+	}
+	if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM |
+			DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) {
+		/* Start the sample generator */
+		spcr |= DAVINCI_MCBSP_SPCR_GRST;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+	}
+
+	if (playback) {
+		/* Enable the transmitter */
+		spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		spcr |= DAVINCI_MCBSP_SPCR_XRST;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+
+		/* wait for any unexpected frame sync error to occur */
+		udelay(100);
+
+		/* Disable the transmitter to clear any outstanding XSYNCERR */
+		spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		spcr &= ~DAVINCI_MCBSP_SPCR_XRST;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+		toggle_clock(dev, playback);
+	}
+
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 2231b2c63f869528504195f202d9585a4f00f143 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 30 Aug 2019 13:38:38 +0300
Subject: ASoC: ti: davinci-i2s: Add S32_LE as support format

ASP/McBSP can support 8/16/20/24/32 bits word in theory. I have only tested
S16_LE and S32_LE, the other formats might not work so only extend the
supported formats with S32_LE for now.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-i2s.c | 8 ++++++--
 1 file changed, 6 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index 92c1bdc69086..f04d9fb5130f 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -612,6 +612,8 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 }
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
+#define DAVINCI_I2S_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+				 SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = {
 	.shutdown	= davinci_i2s_shutdown,
@@ -639,12 +641,14 @@ static struct snd_soc_dai_driver davinci_i2s_dai = {
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = DAVINCI_I2S_FORMATS,
+	},
 	.capture = {
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = DAVINCI_I2S_FORMATS,
+	},
 	.ops = &davinci_i2s_dai_ops,
 
 };
-- 
cgit v1.2.3-59-g8ed1b


From 27b7def93aaba0838eb90e6b3c3110a856bb09fc Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 30 Aug 2019 13:38:40 +0300
Subject: ASoC: ti: edma-pcm: Fix for legacy dma_slave_map based channel lookup

Most of the daVinci devices does not boot with DT. In this case the DMA
channel is looked up with dma_slave_map and for that the chan_names[]
must be configured.

Both McASP and ASP/McBSP uses "tx" and "rx" as channel names, so we can
just do this when the dev->of_node is not valid.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-4-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/edma-pcm.c | 17 ++++++++++++++++-
 1 file changed, 16 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/ti/edma-pcm.c b/sound/soc/ti/edma-pcm.c
index 3ebea1bd15cb..634b040b65f0 100644
--- a/sound/soc/ti/edma-pcm.c
+++ b/sound/soc/ti/edma-pcm.c
@@ -39,7 +39,22 @@ static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = {
 
 int edma_pcm_platform_register(struct device *dev)
 {
-	return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, 0);
+	struct snd_dmaengine_pcm_config *config;
+
+	if (dev->of_node)
+		return devm_snd_dmaengine_pcm_register(dev,
+						&edma_dmaengine_pcm_config, 0);
+
+	config = devm_kzalloc(dev, sizeof(*config), GFP_KERNEL);
+	if (!config)
+		return -ENOMEM;
+
+	*config = edma_dmaengine_pcm_config;
+
+	config->chan_names[0] = "tx";
+	config->chan_names[1] = "rx";
+
+	return devm_snd_dmaengine_pcm_register(dev, config, 0);
 }
 EXPORT_SYMBOL_GPL(edma_pcm_platform_register);
 
-- 
cgit v1.2.3-59-g8ed1b


From 69f34053f883a53e86fa8ff7b99a81d8b5d50dae Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 30 Aug 2019 13:38:41 +0300
Subject: ASoC: ti: davinci-evm: Do not fail if the dai_set_sysclk returns
 -ENOTSUPP

The davinci McBSP (davinci-i2s) driver does not implement the set_sysclk
callback, which is fine and should not be treated as error.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-5-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-evm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
index bfd8d1a03ba7..686b23d7a90d 100644
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -68,7 +68,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 
 	/* set the CPU system clock */
 	ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
-	if (ret < 0)
+	if (ret < 0 && ret != -ENOTSUPP)
 		return ret;
 
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From ca964edf0ddbfec2cb10b3d251d09598e7ca9b13 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 22 Aug 2019 13:36:13 +0200
Subject: ASoC: Intel: Haswell: Adjust machine device private context

Apart from Haswell machines, all other devices have their private data
set to snd_soc_acpi_mach instance.

Changes for HSW/ BDW boards introduced with series:
https://patchwork.kernel.org/cover/10782035/

added support for dai_link platform_name adjustments within card probe
routines. These take for granted private_data points to
snd_soc_acpi_mach whereas for Haswell, it's sst_pdata instead. Change
private context of platform_device - representing machine board - to
address this.

Fixes: e87055d732e3 ("ASoC: Intel: haswell: platform name fixup support")
Fixes: 7e40ddcf974a ("ASoC: Intel: bdw-rt5677: platform name fixup support")
Fixes: 2d067b2807f9 ("ASoC: Intel: broadwell: platform name fixup support")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190822113616.22702-2-cezary.rojewski@intel.com
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/common/sst-acpi.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c
index 0e8e0a7a11df..5854868650b9 100644
--- a/sound/soc/intel/common/sst-acpi.c
+++ b/sound/soc/intel/common/sst-acpi.c
@@ -141,11 +141,12 @@ static int sst_acpi_probe(struct platform_device *pdev)
 	}
 
 	platform_set_drvdata(pdev, sst_acpi);
+	mach->pdata = sst_pdata;
 
 	/* register machine driver */
 	sst_acpi->pdev_mach =
 		platform_device_register_data(dev, mach->drv_name, -1,
-					      sst_pdata, sizeof(*sst_pdata));
+					      mach, sizeof(*mach));
 	if (IS_ERR(sst_acpi->pdev_mach))
 		return PTR_ERR(sst_acpi->pdev_mach);
 
-- 
cgit v1.2.3-59-g8ed1b


From 1fc3e6b2ac3f8125ae54685c600d594d690a268a Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 22 Aug 2019 13:36:14 +0200
Subject: ASoC: Intel: haswell: Simplify device probe

With legacy ADSP private context adjusted, there is no need for double
safety.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190822113616.22702-3-cezary.rojewski@intel.com
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/haswell.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 4d3822cff98c..3dadf9bff796 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -188,18 +188,14 @@ static struct snd_soc_card haswell_rt5640 = {
 static int haswell_audio_probe(struct platform_device *pdev)
 {
 	struct snd_soc_acpi_mach *mach;
-	const char *platform_name = NULL;
 	int ret;
 
 	haswell_rt5640.dev = &pdev->dev;
 
 	/* override plaform name, if required */
 	mach = (&pdev->dev)->platform_data;
-	if (mach) /* extra check since legacy does not pass parameters */
-		platform_name = mach->mach_params.platform;
-
 	ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640,
-						    platform_name);
+						    mach->mach_params.platform);
 	if (ret)
 		return ret;
 
-- 
cgit v1.2.3-59-g8ed1b


From c25e93bba90b3f194c43a37fe2fcdb0727c4ab84 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 22 Aug 2019 13:36:15 +0200
Subject: ASoC: Intel: bdw-rt5677: Simplify device probe

With legacy ADSP private context adjusted, there is no need for double
safety.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190822113616.22702-4-cezary.rojewski@intel.com
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/bdw-rt5677.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index e8e9c3dc82a5..4a4d3353e26d 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -340,7 +340,6 @@ static int bdw_rt5677_probe(struct platform_device *pdev)
 {
 	struct bdw_rt5677_priv *bdw_rt5677;
 	struct snd_soc_acpi_mach *mach;
-	const char *platform_name = NULL;
 	int ret;
 
 	bdw_rt5677_card.dev = &pdev->dev;
@@ -355,11 +354,8 @@ static int bdw_rt5677_probe(struct platform_device *pdev)
 
 	/* override plaform name, if required */
 	mach = (&pdev->dev)->platform_data;
-	if (mach) /* extra check since legacy does not pass parameters */
-		platform_name = mach->mach_params.platform;
-
 	ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5677_card,
-						    platform_name);
+						    mach->mach_params.platform);
 	if (ret)
 		return ret;
 
-- 
cgit v1.2.3-59-g8ed1b


From 54d037d5a466ce518ee4bb81f39b79ac7d843ba6 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 22 Aug 2019 13:36:16 +0200
Subject: ASoC: Intel: broadwell: Simplify device probe

With legacy ADSP private context adjusted, there is no need for double
safety.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190822113616.22702-5-cezary.rojewski@intel.com
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/broadwell.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index ab38ef30dfff..db7e1e87156d 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -270,18 +270,14 @@ static struct snd_soc_card broadwell_rt286 = {
 static int broadwell_audio_probe(struct platform_device *pdev)
 {
 	struct snd_soc_acpi_mach *mach;
-	const char *platform_name = NULL;
 	int ret;
 
 	broadwell_rt286.dev = &pdev->dev;
 
 	/* override plaform name, if required */
 	mach = (&pdev->dev)->platform_data;
-	if (mach) /* extra check since legacy does not pass parameters */
-		platform_name = mach->mach_params.platform;
-
 	ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286,
-						    platform_name);
+						    mach->mach_params.platform);
 	if (ret)
 		return ret;
 
-- 
cgit v1.2.3-59-g8ed1b


From 37638af8135c44e9c9e4b2744eadfd3839efc023 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Thu, 29 Aug 2019 16:42:13 -0500
Subject: ASoC: SOF/Intel: fix selection of SND_INTEL_NHTL

We should only select SND_INTEL_NHLT when ACPI is defined. This was
done for the legacy HDAudio driver but not for DSP-enabled cases,
leading to compilation errors with randconfig.

Fix by aligning on the same solution.

For the Skylake driver this is overkill since there is a top-level
dependency on ACPI, but it doesn't hurt and it's better to have
consistency.

Fixes: 68b953aeb50d9 ('ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number')
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190829214213.11653-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/Kconfig     | 2 +-
 sound/soc/sof/intel/Kconfig | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index a3ec17fd63cd..01c99750212a 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON
 	select SND_SOC_INTEL_SST
 	select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
 	select SND_SOC_ACPI_INTEL_MATCH
-	select SND_INTEL_NHLT
+	select SND_INTEL_NHLT if ACPI
 	help
 	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
 	  GeminiLake or CannonLake platform with the DSP enabled in the BIOS
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 889b6202d054..479ba249e219 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA
 	tristate
 	select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK
 	select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC
-	select SND_INTEL_NHLT
+	select SND_INTEL_NHLT if ACPI
 	help
 	  This option is not user-selectable but automagically handled by
 	  'select' statements at a higher level
-- 
cgit v1.2.3-59-g8ed1b


From 96ed76983307a8a24400495e4e6453607e765fcf Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Fri, 30 Aug 2019 15:42:37 +0800
Subject: ASoC: mediatek: mt8183: move private structure

Move private structure to the beginning of file to declare earlier
so that most functions can see it.

Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190830074240.195166-2-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../mt8183/mt8183-mt6358-ts3a227-max98357.c        | 32 +++++++++++-----------
 1 file changed, 16 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 560fcc5923a4..1d4a1600f768 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -17,6 +17,22 @@
 
 static struct snd_soc_jack headset_jack;
 
+enum PINCTRL_PIN_STATE {
+	PIN_STATE_DEFAULT = 0,
+	PIN_TDM_OUT_ON,
+	PIN_TDM_OUT_OFF,
+	PIN_STATE_MAX
+};
+
+static const char * const mt8183_pin_str[PIN_STATE_MAX] = {
+	"default", "aud_tdm_out_on", "aud_tdm_out_off",
+};
+
+struct mt8183_mt6358_ts3a227_max98357_priv {
+	struct pinctrl *pinctrl;
+	struct pinctrl_state *pin_states[PIN_STATE_MAX];
+};
+
 static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
 				       struct snd_pcm_hw_params *params)
 {
@@ -46,22 +62,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-enum PINCTRL_PIN_STATE {
-	PIN_STATE_DEFAULT = 0,
-	PIN_TDM_OUT_ON,
-	PIN_TDM_OUT_OFF,
-	PIN_STATE_MAX
-};
-
-static const char * const mt8183_pin_str[PIN_STATE_MAX] = {
-	"default", "aud_tdm_out_on", "aud_tdm_out_off",
-};
-
-struct mt8183_mt6358_ts3a227_max98357_priv {
-	struct pinctrl *pinctrl;
-	struct pinctrl_state *pin_states[PIN_STATE_MAX];
-};
-
 static int
 mt8183_mt6358_ts3a227_max98357_bt_sco_startup(
 	struct snd_pcm_substream *substream)
-- 
cgit v1.2.3-59-g8ed1b


From 95d779b4b8e7a62d53c7afeba45207089dcee4c8 Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Fri, 30 Aug 2019 15:42:38 +0800
Subject: ASoC: mediatek: mt8183: move headset jack to card-specific storage

Move headset jack descriptor from module global scope to card-specific
storage to make its ownership more explicit.

Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190830074240.195166-3-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 1d4a1600f768..a7ad41b6e885 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -15,8 +15,6 @@
 #include "mt8183-afe-common.h"
 #include "../../codecs/ts3a227e.h"
 
-static struct snd_soc_jack headset_jack;
-
 enum PINCTRL_PIN_STATE {
 	PIN_STATE_DEFAULT = 0,
 	PIN_TDM_OUT_ON,
@@ -31,6 +29,7 @@ static const char * const mt8183_pin_str[PIN_STATE_MAX] = {
 struct mt8183_mt6358_ts3a227_max98357_priv {
 	struct pinctrl *pinctrl;
 	struct pinctrl_state *pin_states[PIN_STATE_MAX];
+	struct snd_soc_jack headset_jack;
 };
 
 static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
@@ -410,6 +409,8 @@ static int
 mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *component)
 {
 	int ret;
+	struct mt8183_mt6358_ts3a227_max98357_priv *priv =
+			snd_soc_card_get_drvdata(component->card);
 
 	/* Enable Headset and 4 Buttons Jack detection */
 	ret = snd_soc_card_jack_new(&mt8183_mt6358_ts3a227_max98357_card,
@@ -417,12 +418,12 @@ mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *component)
 				    SND_JACK_HEADSET |
 				    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 				    SND_JACK_BTN_2 | SND_JACK_BTN_3,
-				    &headset_jack,
+				    &priv->headset_jack,
 				    NULL, 0);
 	if (ret)
 		return ret;
 
-	ret = ts3a227e_enable_jack_detect(component, &headset_jack);
+	ret = ts3a227e_enable_jack_detect(component, &priv->headset_jack);
 
 	return ret;
 }
-- 
cgit v1.2.3-59-g8ed1b


From fc347483b87f029c1bc18a8492623ca21dcd7607 Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Fri, 30 Aug 2019 15:42:39 +0800
Subject: ASoC: mediatek: mt8183: remove forward declaration of headset_init

The forward declaration of mt8183_mt6358_ts3a227_max98357_headset_init
is for cyclic dependency between card, headset_dev, and headset_init.

It used to be:
- card depends on headset_dev
- headset_dev depends on headset_init
- headset_init depends on card

Commit a962a809e5e4 ("ASoC: mediatek: mt8183: make headset codec
optional") removed the cyclic dependency.

Thus, it is safe to remove the forward declaration.

Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190830074240.195166-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 13 +++++--------
 1 file changed, 5 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index a7ad41b6e885..30a7735c4af3 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -390,14 +390,6 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = {
 	},
 };
 
-static int
-mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *cpnt);
-
-static struct snd_soc_aux_dev mt8183_mt6358_ts3a227_max98357_headset_dev = {
-	.dlc = COMP_EMPTY(),
-	.init = mt8183_mt6358_ts3a227_max98357_headset_init,
-};
-
 static struct snd_soc_card mt8183_mt6358_ts3a227_max98357_card = {
 	.name = "mt8183_mt6358_ts3a227_max98357",
 	.owner = THIS_MODULE,
@@ -428,6 +420,11 @@ mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *component)
 	return ret;
 }
 
+static struct snd_soc_aux_dev mt8183_mt6358_ts3a227_max98357_headset_dev = {
+	.dlc = COMP_EMPTY(),
+	.init = mt8183_mt6358_ts3a227_max98357_headset_init,
+};
+
 static int
 mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev)
 {
-- 
cgit v1.2.3-59-g8ed1b


From e5b3ae3e8c4772106713698075a87bff1e86e710 Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Fri, 30 Aug 2019 15:42:40 +0800
Subject: ASoC: mediatek: mt8183: fix space issues

Fix space issues:
- No space before ';'.
- No trailing blank line.

Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190830074240.195166-5-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 30a7735c4af3..bb9cdc0d6552 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -471,7 +471,7 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev)
 		return PTR_ERR(priv->pinctrl);
 	}
 
-	for (i = 0 ; i < PIN_STATE_MAX ; i++) {
+	for (i = 0; i < PIN_STATE_MAX; i++) {
 		priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl,
 							   mt8183_pin_str[i]);
 		if (IS_ERR(priv->pin_states[i])) {
@@ -526,4 +526,3 @@ MODULE_DESCRIPTION("MT8183-MT6358-TS3A227-MAX98357 ALSA SoC machine driver");
 MODULE_AUTHOR("Shunli Wang <shunli.wang@mediatek.com>");
 MODULE_LICENSE("GPL v2");
 MODULE_ALIAS("mt8183_mt6358_ts3a227_max98357 soc card");
-
-- 
cgit v1.2.3-59-g8ed1b


From 9a7c9fe1203eb360b3a01b65a18bcd0de6670c53 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 23 Aug 2019 09:58:37 +0900
Subject: ASoC: soc-core: add soc_rtd_free()

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

soc_rtd_init() was soc_post_component_init(), but there was no
its paired soc_post_component_free(), but it is done at
soc_remove_link_dais().
This means it is difficult to find related code.

This patch adds soc_rtd_free() which is paired soc_rtd_init().
soc_rtd_xxx() will be more cleanuped in the future.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tva87lby.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 17 ++++++++++++-----
 1 file changed, 12 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3c087b478398..3754a08baf62 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -975,17 +975,15 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 	dai->probed = 0;
 }
 
+static void soc_rtd_free(struct snd_soc_pcm_runtime *rtd); /* remove me */
 static void soc_remove_link_dais(struct snd_soc_card *card,
 		struct snd_soc_pcm_runtime *rtd, int order)
 {
 	int i;
 	struct snd_soc_dai *codec_dai;
 
-	/* unregister the rtd device */
-	if (rtd->dev_registered) {
-		device_unregister(rtd->dev);
-		rtd->dev_registered = 0;
-	}
+	/* finalize rtd device */
+	soc_rtd_free(rtd);
 
 	/* remove the CODEC DAI */
 	for_each_rtd_codec_dai(rtd, i, codec_dai)
@@ -1340,6 +1338,15 @@ err_probe:
 	return ret;
 }
 
+static void soc_rtd_free(struct snd_soc_pcm_runtime *rtd)
+{
+	if (rtd->dev_registered) {
+		/* we don't need to call kfree() for rtd->dev */
+		device_unregister(rtd->dev);
+		rtd->dev_registered = 0;
+	}
+}
+
 static void soc_rtd_release(struct device *dev)
 {
 	kfree(dev);
-- 
cgit v1.2.3-59-g8ed1b


From ffd60fba19d9752f553aac367cd40362011ab6c9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 23 Aug 2019 09:58:42 +0900
Subject: ASoC: soc-core: move soc_probe_component() position

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

soc_probe_comonent() has paired soc_remove_comonent(),
but, these are implemented at different place.
So it is difficult to confirm code.
This patch moves soc_probe_component() next to
soc_remove_component().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgps7lbt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 261 +++++++++++++++++++++++++--------------------------
 1 file changed, 130 insertions(+), 131 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3754a08baf62..8fa1cfcc6f17 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -938,6 +938,41 @@ _err_defer:
 	return -EPROBE_DEFER;
 }
 
+static void soc_set_of_name_prefix(struct snd_soc_component *component)
+{
+	struct device_node *of_node = soc_component_to_node(component);
+	const char *str;
+	int ret;
+
+	ret = of_property_read_string(of_node, "sound-name-prefix", &str);
+	if (!ret)
+		component->name_prefix = str;
+}
+
+static void soc_set_name_prefix(struct snd_soc_card *card,
+				struct snd_soc_component *component)
+{
+	int i;
+
+	for (i = 0; i < card->num_configs && card->codec_conf; i++) {
+		struct snd_soc_codec_conf *map = &card->codec_conf[i];
+		struct device_node *of_node = soc_component_to_node(component);
+
+		if (map->of_node && of_node != map->of_node)
+			continue;
+		if (map->dev_name && strcmp(component->name, map->dev_name))
+			continue;
+		component->name_prefix = map->name_prefix;
+		return;
+	}
+
+	/*
+	 * If there is no configuration table or no match in the table,
+	 * check if a prefix is provided in the node
+	 */
+	soc_set_of_name_prefix(component);
+}
+
 static void soc_cleanup_component(struct snd_soc_component *component)
 {
 	snd_soc_component_set_jack(component, NULL, NULL);
@@ -958,6 +993,101 @@ static void soc_remove_component(struct snd_soc_component *component)
 	soc_cleanup_component(component);
 }
 
+static int soc_probe_component(struct snd_soc_card *card,
+			       struct snd_soc_component *component)
+{
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
+	struct snd_soc_dai *dai;
+	int ret;
+
+	if (!strcmp(component->name, "snd-soc-dummy"))
+		return 0;
+
+	if (component->card) {
+		if (component->card != card) {
+			dev_err(component->dev,
+				"Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n",
+				card->name, component->card->name);
+			return -ENODEV;
+		}
+		return 0;
+	}
+
+	ret = snd_soc_component_module_get_when_probe(component);
+	if (ret < 0)
+		return ret;
+
+	component->card = card;
+	dapm->card = card;
+	INIT_LIST_HEAD(&dapm->list);
+	soc_set_name_prefix(card, component);
+
+	soc_init_component_debugfs(component);
+
+	ret = snd_soc_dapm_new_controls(dapm,
+					component->driver->dapm_widgets,
+					component->driver->num_dapm_widgets);
+
+	if (ret != 0) {
+		dev_err(component->dev,
+			"Failed to create new controls %d\n", ret);
+		goto err_probe;
+	}
+
+	for_each_component_dais(component, dai) {
+		ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
+		if (ret != 0) {
+			dev_err(component->dev,
+				"Failed to create DAI widgets %d\n", ret);
+			goto err_probe;
+		}
+	}
+
+	ret = snd_soc_component_probe(component);
+	if (ret < 0) {
+		dev_err(component->dev,
+			"ASoC: failed to probe component %d\n", ret);
+		goto err_probe;
+	}
+	WARN(dapm->idle_bias_off &&
+	     dapm->bias_level != SND_SOC_BIAS_OFF,
+	     "codec %s can not start from non-off bias with idle_bias_off==1\n",
+	     component->name);
+
+	/* machine specific init */
+	if (component->init) {
+		ret = component->init(component);
+		if (ret < 0) {
+			dev_err(component->dev,
+				"Failed to do machine specific init %d\n", ret);
+			goto err_probe;
+		}
+	}
+
+	ret = snd_soc_add_component_controls(component,
+					     component->driver->controls,
+					     component->driver->num_controls);
+	if (ret < 0)
+		goto err_probe;
+
+	ret = snd_soc_dapm_add_routes(dapm,
+				      component->driver->dapm_routes,
+				      component->driver->num_dapm_routes);
+	if (ret < 0)
+		goto err_probe;
+
+	list_add(&dapm->list, &card->dapm_list);
+	/* see for_each_card_components */
+	list_add(&component->card_list, &card->component_dev_list);
+
+err_probe:
+	if (ret < 0)
+		soc_cleanup_component(component);
+
+	return ret;
+}
+
 static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 {
 	int err;
@@ -1207,137 +1337,6 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link);
 
-static void soc_set_of_name_prefix(struct snd_soc_component *component)
-{
-	struct device_node *component_of_node = soc_component_to_node(component);
-	const char *str;
-	int ret;
-
-	ret = of_property_read_string(component_of_node, "sound-name-prefix",
-				      &str);
-	if (!ret)
-		component->name_prefix = str;
-}
-
-static void soc_set_name_prefix(struct snd_soc_card *card,
-				struct snd_soc_component *component)
-{
-	int i;
-
-	for (i = 0; i < card->num_configs && card->codec_conf; i++) {
-		struct snd_soc_codec_conf *map = &card->codec_conf[i];
-		struct device_node *component_of_node = soc_component_to_node(component);
-
-		if (map->of_node && component_of_node != map->of_node)
-			continue;
-		if (map->dev_name && strcmp(component->name, map->dev_name))
-			continue;
-		component->name_prefix = map->name_prefix;
-		return;
-	}
-
-	/*
-	 * If there is no configuration table or no match in the table,
-	 * check if a prefix is provided in the node
-	 */
-	soc_set_of_name_prefix(component);
-}
-
-static int soc_probe_component(struct snd_soc_card *card,
-	struct snd_soc_component *component)
-{
-	struct snd_soc_dapm_context *dapm =
-			snd_soc_component_get_dapm(component);
-	struct snd_soc_dai *dai;
-	int ret;
-
-	if (!strcmp(component->name, "snd-soc-dummy"))
-		return 0;
-
-	if (component->card) {
-		if (component->card != card) {
-			dev_err(component->dev,
-				"Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n",
-				card->name, component->card->name);
-			return -ENODEV;
-		}
-		return 0;
-	}
-
-	ret = snd_soc_component_module_get_when_probe(component);
-	if (ret < 0)
-		return ret;
-
-	component->card = card;
-	dapm->card = card;
-	INIT_LIST_HEAD(&dapm->list);
-	soc_set_name_prefix(card, component);
-
-	soc_init_component_debugfs(component);
-
-	ret = snd_soc_dapm_new_controls(dapm,
-					component->driver->dapm_widgets,
-					component->driver->num_dapm_widgets);
-
-	if (ret != 0) {
-		dev_err(component->dev,
-			"Failed to create new controls %d\n", ret);
-		goto err_probe;
-	}
-
-	for_each_component_dais(component, dai) {
-		ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
-		if (ret != 0) {
-			dev_err(component->dev,
-				"Failed to create DAI widgets %d\n", ret);
-			goto err_probe;
-		}
-	}
-
-	ret = snd_soc_component_probe(component);
-	if (ret < 0) {
-		dev_err(component->dev,
-			"ASoC: failed to probe component %d\n", ret);
-		goto err_probe;
-	}
-	WARN(dapm->idle_bias_off &&
-	     dapm->bias_level != SND_SOC_BIAS_OFF,
-	     "codec %s can not start from non-off bias with idle_bias_off==1\n",
-	     component->name);
-
-	/* machine specific init */
-	if (component->init) {
-		ret = component->init(component);
-		if (ret < 0) {
-			dev_err(component->dev,
-				"Failed to do machine specific init %d\n", ret);
-			goto err_probe;
-		}
-	}
-
-	ret = snd_soc_add_component_controls(component,
-					     component->driver->controls,
-					     component->driver->num_controls);
-	if (ret < 0)
-		goto err_probe;
-
-	ret = snd_soc_dapm_add_routes(dapm,
-				      component->driver->dapm_routes,
-				      component->driver->num_dapm_routes);
-	if (ret < 0)
-		goto err_probe;
-
-	list_add(&dapm->list, &card->dapm_list);
-	/* see for_each_card_components */
-	list_add(&component->card_list, &card->component_dev_list);
-
-err_probe:
-	if (ret < 0)
-		soc_cleanup_component(component);
-
-	return ret;
-}
-
 static void soc_rtd_free(struct snd_soc_pcm_runtime *rtd)
 {
 	if (rtd->dev_registered) {
-- 
cgit v1.2.3-59-g8ed1b


From b614beafa495c7f6fbc15cb6977e3fe48beea1e5 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 23 Aug 2019 09:58:47 +0900
Subject: ASoC: soc-core: dapm related setup at one place

Current ASoC setups some dapm related member at
snd_soc_component_initialize() which is called when component was
registered, and setups remaining member at soc_probe_component()
which is called when component was probed.
This kind of setup separation is no meanings, and it is very
difficult to read and confusable.
This patch setups all dapm settings at one place.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r25c7lbo.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 21 +++++++++------------
 1 file changed, 9 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 8fa1cfcc6f17..21c005a044e8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1019,12 +1019,19 @@ static int soc_probe_component(struct snd_soc_card *card,
 		return ret;
 
 	component->card = card;
-	dapm->card = card;
-	INIT_LIST_HEAD(&dapm->list);
 	soc_set_name_prefix(card, component);
 
 	soc_init_component_debugfs(component);
 
+	INIT_LIST_HEAD(&dapm->list);
+	dapm->card		= card;
+	dapm->dev		= component->dev;
+	dapm->component		= component;
+	dapm->bias_level	= SND_SOC_BIAS_OFF;
+	dapm->idle_bias_off	= !component->driver->idle_bias_on;
+	dapm->suspend_bias_off	= component->driver->suspend_bias_off;
+	list_add(&dapm->list, &card->dapm_list);
+
 	ret = snd_soc_dapm_new_controls(dapm,
 					component->driver->dapm_widgets,
 					component->driver->num_dapm_widgets);
@@ -1077,7 +1084,6 @@ static int soc_probe_component(struct snd_soc_card *card,
 	if (ret < 0)
 		goto err_probe;
 
-	list_add(&dapm->list, &card->dapm_list);
 	/* see for_each_card_components */
 	list_add(&component->card_list, &card->component_dev_list);
 
@@ -2649,8 +2655,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dai);
 static int snd_soc_component_initialize(struct snd_soc_component *component,
 	const struct snd_soc_component_driver *driver, struct device *dev)
 {
-	struct snd_soc_dapm_context *dapm;
-
 	INIT_LIST_HEAD(&component->dai_list);
 	INIT_LIST_HEAD(&component->dobj_list);
 	INIT_LIST_HEAD(&component->card_list);
@@ -2665,13 +2669,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	component->dev = dev;
 	component->driver = driver;
 
-	dapm = snd_soc_component_get_dapm(component);
-	dapm->dev = dev;
-	dapm->component = component;
-	dapm->bias_level = SND_SOC_BIAS_OFF;
-	dapm->idle_bias_off = !driver->idle_bias_on;
-	dapm->suspend_bias_off = driver->suspend_bias_off;
-
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From 95c267dd20431f0eb54ca204bd73a7d85c532a37 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 23 Aug 2019 09:58:52 +0900
Subject: ASoC: soc-core: add snd_soc_dapm_init()

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

soc-dapm has snd_soc_dapm_free() which cleanups debugfs, widgets, list.
But, there is no paired initialize function.
This patch adds snd_soc_dapm_init() and initilaizing dapm

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pnkw7lbj.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dapm.h |  3 +++
 sound/soc/soc-core.c     | 14 ++------------
 sound/soc/soc-dapm.c     | 21 +++++++++++++++++++++
 3 files changed, 26 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 2aa73d6dd7be..dd993dd29229 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -416,6 +416,9 @@ int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream,
 /* dapm path setup */
 int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
+void snd_soc_dapm_init(struct snd_soc_dapm_context *dapm,
+		       struct snd_soc_card *card,
+		       struct snd_soc_component *component);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
 int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 21c005a044e8..8e831ae59eb8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1023,14 +1023,7 @@ static int soc_probe_component(struct snd_soc_card *card,
 
 	soc_init_component_debugfs(component);
 
-	INIT_LIST_HEAD(&dapm->list);
-	dapm->card		= card;
-	dapm->dev		= component->dev;
-	dapm->component		= component;
-	dapm->bias_level	= SND_SOC_BIAS_OFF;
-	dapm->idle_bias_off	= !component->driver->idle_bias_on;
-	dapm->suspend_bias_off	= component->driver->suspend_bias_off;
-	list_add(&dapm->list, &card->dapm_list);
+	snd_soc_dapm_init(dapm, card, component);
 
 	ret = snd_soc_dapm_new_controls(dapm,
 					component->driver->dapm_widgets,
@@ -1937,10 +1930,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
 
-	card->dapm.bias_level = SND_SOC_BIAS_OFF;
-	card->dapm.dev = card->dev;
-	card->dapm.card = card;
-	list_add(&card->dapm.list, &card->dapm_list);
+	snd_soc_dapm_init(&card->dapm, card, NULL);
 
 	/* check whether any platform is ignore machine FE and using topology */
 	soc_check_tplg_fes(card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 10819b3e0b98..b6378f025836 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4717,6 +4717,27 @@ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm)
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
 
+void snd_soc_dapm_init(struct snd_soc_dapm_context *dapm,
+		       struct snd_soc_card *card,
+		       struct snd_soc_component *component)
+{
+	dapm->card		= card;
+	dapm->component		= component;
+	dapm->bias_level	= SND_SOC_BIAS_OFF;
+
+	if (component) {
+		dapm->dev		= component->dev;
+		dapm->idle_bias_off	= !component->driver->idle_bias_on,
+		dapm->suspend_bias_off	= component->driver->suspend_bias_off;
+	} else {
+		dapm->dev		= card->dev;
+	}
+
+	INIT_LIST_HEAD(&dapm->list);
+	list_add(&dapm->list, &card->dapm_list);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_init);
+
 static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_card *card = dapm->card;
-- 
cgit v1.2.3-59-g8ed1b


From 6fb035502956ad1f338ca61c057653d5372ffd8c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 23 Aug 2019 09:58:58 +0900
Subject: ASoC: soc-core: move soc_probe_link_components() position

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

soc_probe_link_components() has paired soc_remove_link_components(),
but, these are implemented at different place.
So it is difficult to confirm code.
This patch moves soc_probe_link_components() next to
soc_remove_link_components().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o90g7lbd.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 40 ++++++++++++++++++++--------------------
 1 file changed, 20 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 8e831ae59eb8..2a166abaade1 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1135,6 +1135,26 @@ static void soc_remove_link_components(struct snd_soc_card *card,
 	}
 }
 
+static int soc_probe_link_components(struct snd_soc_card *card,
+				     struct snd_soc_pcm_runtime *rtd, int order)
+{
+	struct snd_soc_component *component;
+	struct snd_soc_rtdcom_list *rtdcom;
+	int ret;
+
+	for_each_rtdcom(rtd, rtdcom) {
+		component = rtdcom->component;
+
+		if (component->driver->probe_order == order) {
+			ret = soc_probe_component(card, component);
+			if (ret < 0)
+				return ret;
+		}
+	}
+
+	return 0;
+}
+
 static void soc_remove_dai_links(struct snd_soc_card *card)
 {
 	int order;
@@ -1379,26 +1399,6 @@ static int soc_rtd_init(struct snd_soc_pcm_runtime *rtd, const char *name)
 	return 0;
 }
 
-static int soc_probe_link_components(struct snd_soc_card *card,
-				     struct snd_soc_pcm_runtime *rtd, int order)
-{
-	struct snd_soc_component *component;
-	struct snd_soc_rtdcom_list *rtdcom;
-	int ret;
-
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
-
-		if (component->driver->probe_order == order) {
-			ret = soc_probe_component(card, component);
-			if (ret < 0)
-				return ret;
-		}
-	}
-
-	return 0;
-}
-
 static int soc_probe_dai(struct snd_soc_dai *dai, int order)
 {
 	int ret;
-- 
cgit v1.2.3-59-g8ed1b


From 39c66b6ef6f3aaf5bac61555db70ffb2864161c6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 3 Sep 2019 17:15:54 +0900
Subject: ASoC: simple-card: indicate rebind issue

ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform
were unbinded and re-binded again.
But, Simple Card might can't rebind again if user do for example

	unbind CPU or Codec driver
	bind   CPU or Codec driver

Because Simple Card is still pointing old/unbinded
CPU or Codec driver's DAI name at dlc->dai_name.

To avoid this issue, it needs to alloc memory and keep DAI name
even though if CPU or Codec driver was unbinded.
Or, always do unbind/bind at Sound Card.

For now, this patch indicates this issue as FIXME.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tv9tu75x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card.c | 20 +++++++++++++++++++-
 1 file changed, 19 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 0eac3bcb9736..fc9c753db8dd 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -46,7 +46,25 @@ static int asoc_simple_parse_dai(struct device_node *node,
 	if (ret)
 		return ret;
 
-	/* Get dai->name */
+	/*
+	 * FIXME
+	 *
+	 * Here, dlc->dai_name is pointer to CPU/Codec DAI name.
+	 * If user unbinded CPU or Codec driver, but not for Sound Card,
+	 * dlc->dai_name is keeping unbinded CPU or Codec
+	 * driver's pointer.
+	 *
+	 * If user re-bind CPU or Codec driver again, ALSA SoC will try
+	 * to rebind Card via snd_soc_try_rebind_card(), but because of
+	 * above reason, it might can't bind Sound Card.
+	 * Because Sound Card is pointing to released dai_name pointer.
+	 *
+	 * To avoid this rebind Card issue,
+	 * 1) It needs to alloc memory to keep dai_name eventhough
+	 *    CPU or Codec driver was unbinded, or
+	 * 2) user need to rebind Sound Card everytime
+	 *    if he unbinded CPU or Codec.
+	 */
 	ret = snd_soc_of_get_dai_name(node, &dlc->dai_name);
 	if (ret < 0)
 		return ret;
-- 
cgit v1.2.3-59-g8ed1b


From 157ab71286e6dd71676052275ab5dfa3955c5cdb Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 3 Sep 2019 17:16:05 +0900
Subject: ASoC: audio-graph: indicate rebind issue

ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform
were unbinded and re-binded again.
But, Audio Graph Card might can't rebind again if user do for example

	unbind CPU or Codec driver
	bind   CPU or Codec driver

Because Audio Graph Card is still pointing old/unbinded
CPU or Codec driver's DAI name at dlc->dai_name.

To avoid this issue, it needs to alloc memory and keep DAI name
even though if CPU or Codec driver was unbinded.
Or, always do unbind/bind at Sound Card.

For now, this patch indicates this issue as FIXME.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgpdu75m.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card.c | 19 +++++++++++++++++++
 1 file changed, 19 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 288df245b2f0..6007e6305735 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -129,6 +129,25 @@ static int asoc_simple_parse_dai(struct device_node *ep,
 	args.args[0]	= graph_get_dai_id(ep);
 	args.args_count	= (of_graph_get_endpoint_count(node) > 1);
 
+	/*
+	 * FIXME
+	 *
+	 * Here, dlc->dai_name is pointer to CPU/Codec DAI name.
+	 * If user unbinded CPU or Codec driver, but not for Sound Card,
+	 * dlc->dai_name is keeping unbinded CPU or Codec
+	 * driver's pointer.
+	 *
+	 * If user re-bind CPU or Codec driver again, ALSA SoC will try
+	 * to rebind Card via snd_soc_try_rebind_card(), but because of
+	 * above reason, it might can't bind Sound Card.
+	 * Because Sound Card is pointing to released dai_name pointer.
+	 *
+	 * To avoid this rebind Card issue,
+	 * 1) It needs to alloc memory to keep dai_name eventhough
+	 *    CPU or Codec driver was unbinded, or
+	 * 2) user need to rebind Sound Card everytime
+	 *    if he unbinded CPU or Codec.
+	 */
 	ret = snd_soc_get_dai_name(&args, &dlc->dai_name);
 	if (ret < 0)
 		return ret;
-- 
cgit v1.2.3-59-g8ed1b


From 6dd567dc964878c04269a44114ef13693712edf1 Mon Sep 17 00:00:00 2001
From: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Date: Wed, 4 Sep 2019 01:53:20 +0900
Subject: ASoC: es8316: add clock control of MCLK

This patch introduce clock property for MCLK master freq control.
Driver will set rate of MCLK master if set_sysclk is called and
changing sysclk by board driver.

[Modified slightly to apply without an earlier patch in the series due
to context diffs -- broonie]

Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190903165322.20791-2-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8316.c | 36 +++++++++++++++++++++++++++++++++++-
 1 file changed, 35 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 6db002cc2058..6248b01ca049 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -9,6 +9,7 @@
 
 #include <linux/module.h>
 #include <linux/acpi.h>
+#include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/mod_devicetable.h>
@@ -33,6 +34,7 @@ static const unsigned int supported_mclk_lrck_ratios[] = {
 
 struct es8316_priv {
 	struct mutex lock;
+	struct clk *mclk;
 	struct regmap *regmap;
 	struct snd_soc_component *component;
 	struct snd_soc_jack *jack;
@@ -360,7 +362,7 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_component *component = codec_dai->component;
 	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
-	int i;
+	int i, ret;
 	int count = 0;
 
 	es8316->sysclk = freq;
@@ -368,6 +370,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	if (freq == 0)
 		return 0;
 
+	if (es8316->mclk) {
+		ret = clk_set_rate(es8316->mclk, freq);
+		if (ret)
+			return ret;
+	}
+
 	/* Limit supported sample rates to ones that can be autodetected
 	 * by the codec running in slave mode.
 	 */
@@ -697,9 +705,26 @@ static int es8316_set_jack(struct snd_soc_component *component,
 static int es8316_probe(struct snd_soc_component *component)
 {
 	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
+	int ret;
 
 	es8316->component = component;
 
+	es8316->mclk = devm_clk_get(component->dev, "mclk");
+	if (PTR_ERR(es8316->mclk) == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+	if (IS_ERR(es8316->mclk)) {
+		dev_err(component->dev, "clock is invalid, ignored\n");
+		es8316->mclk = NULL;
+	}
+
+	if (es8316->mclk) {
+		ret = clk_prepare_enable(es8316->mclk);
+		if (ret) {
+			dev_err(component->dev, "unable to enable clock\n");
+			return ret;
+		}
+	}
+
 	/* Reset codec and enable current state machine */
 	snd_soc_component_write(component, ES8316_RESET, 0x3f);
 	usleep_range(5000, 5500);
@@ -722,8 +747,17 @@ static int es8316_probe(struct snd_soc_component *component)
 	return 0;
 }
 
+static void es8316_remove(struct snd_soc_component *component)
+{
+	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
+
+	if (es8316->mclk)
+		clk_disable_unprepare(es8316->mclk);
+}
+
 static const struct snd_soc_component_driver soc_component_dev_es8316 = {
 	.probe			= es8316_probe,
+	.remove			= es8316_remove,
 	.set_jack		= es8316_set_jack,
 	.controls		= es8316_snd_controls,
 	.num_controls		= ARRAY_SIZE(es8316_snd_controls),
-- 
cgit v1.2.3-59-g8ed1b


From dfafc1822f6826c9d250223f59ce8c3b227866a6 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Wed, 4 Sep 2019 15:48:33 +0800
Subject: ASoC: amd: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190904074833.23572-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/acp-pcm-dma.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index d26653f81416..52225b4b6382 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -1251,8 +1251,7 @@ static int acp_audio_probe(struct platform_device *pdev)
 	if (!audio_drv_data)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res);
+	audio_drv_data->acp_mmio = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(audio_drv_data->acp_mmio))
 		return PTR_ERR(audio_drv_data->acp_mmio);
 
-- 
cgit v1.2.3-59-g8ed1b


From f295495ec657c5fb2cff355456c2a20c4c945d93 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Wed, 4 Sep 2019 16:25:07 +0800
Subject: ASoC: ep93xx: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190904082507.24300-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/cirrus/ep93xx-ac97.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index 84c967fcab6b..e21eaa1893d1 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -362,7 +362,6 @@ static const struct snd_soc_component_driver ep93xx_ac97_component = {
 static int ep93xx_ac97_probe(struct platform_device *pdev)
 {
 	struct ep93xx_ac97_info *info;
-	struct resource *res;
 	int irq;
 	int ret;
 
@@ -370,8 +369,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev)
 	if (!info)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	info->regs = devm_ioremap_resource(&pdev->dev, res);
+	info->regs = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(info->regs))
 		return PTR_ERR(info->regs);
 
-- 
cgit v1.2.3-59-g8ed1b


From 2f302d476c960fdf8481399a46b8df92408d06e2 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Wed, 4 Sep 2019 16:34:12 +0800
Subject: ASoC: sirf-audio: use devm_platform_ioremap_resource() to simplify
 code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190904083412.18700-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sirf-audio-codec.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
index 9009a7407b7a..a061d78473ac 100644
--- a/sound/soc/codecs/sirf-audio-codec.c
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -459,7 +459,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
 	int ret;
 	struct sirf_audio_codec *sirf_audio_codec;
 	void __iomem *base;
-	struct resource *mem_res;
 
 	sirf_audio_codec = devm_kzalloc(&pdev->dev,
 		sizeof(struct sirf_audio_codec), GFP_KERNEL);
@@ -468,8 +467,7 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(pdev, sirf_audio_codec);
 
-	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_ioremap_resource(&pdev->dev, mem_res);
+	base = devm_platform_ioremap_resource(pdev, 0);
 	if (IS_ERR(base))
 		return PTR_ERR(base);
 
-- 
cgit v1.2.3-59-g8ed1b


From a813d0e8884e514e6e1b28ad3b22f7658d446b16 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Wed, 4 Sep 2019 16:39:09 +0800
Subject: ASoC: tegra: use devm_platform_ioremap_resource() to simplify code

Use devm_platform_ioremap_resource() to simplify the code a bit.
This is detected by coccinelle.

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190904083909.18804-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/tegra/tegra30_ahub.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
index 952381260dc3..635eacbd28d4 100644
--- a/sound/soc/tegra/tegra30_ahub.c
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -511,7 +511,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev)
 	const struct tegra30_ahub_soc_data *soc_data;
 	struct reset_control *rst;
 	int i;
-	struct resource *res0, *res1;
+	struct resource *res0;
 	void __iomem *regs_apbif, *regs_ahub;
 	int ret = 0;
 
@@ -587,8 +587,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev)
 	}
 	regcache_cache_only(ahub->regmap_apbif, true);
 
-	res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1);
-	regs_ahub = devm_ioremap_resource(&pdev->dev, res1);
+	regs_ahub = devm_platform_ioremap_resource(pdev, 1);
 	if (IS_ERR(regs_ahub))
 		return PTR_ERR(regs_ahub);
 
-- 
cgit v1.2.3-59-g8ed1b


From 04f770d968344ca95b55cf7b50452888ec5440e7 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:14:35 +0900
Subject: ASoC: soc-core: add comment to jack at soc_remove_component()

Basically, driver which setups snd_soc_component_set_jack() need
to release it by themselves. But, as framework level robustness,
soc_remove_component() also releases it.

To avoid code reader confuse, this patch makes it clarify.

This patch makes it clarify.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875zm8q5n8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2a166abaade1..05a2aff843aa 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -975,7 +975,9 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
 
 static void soc_cleanup_component(struct snd_soc_component *component)
 {
+	/* For framework level robustness */
 	snd_soc_component_set_jack(component, NULL, NULL);
+
 	list_del(&component->card_list);
 	snd_soc_dapm_free(snd_soc_component_get_dapm(component));
 	soc_cleanup_component_debugfs(component);
-- 
cgit v1.2.3-59-g8ed1b


From 1ab1b340f037760ec9944a9a4911e85f29fc6e2c Mon Sep 17 00:00:00 2001
From: Bjorn Andersson <bjorn.andersson@linaro.org>
Date: Wed, 4 Sep 2019 21:03:06 -0700
Subject: ASoC: qcom: common: Include link-name in error messages

Reading out the link-name earlier and including it in the various error
messages makes it much more convenient to figure out what links have
unmet dependencies.

Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org>
Link: https://lore.kernel.org/r/20190905040306.21399-1-bjorn.andersson@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/common.c | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 2c7348ddbbb3..6c20bdd850f3 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -53,12 +53,18 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 		link->num_cpus		= 1;
 		link->num_platforms	= 1;
 
+		ret = of_property_read_string(np, "link-name", &link->name);
+		if (ret) {
+			dev_err(card->dev, "error getting codec dai_link name\n");
+			goto err;
+		}
+
 		cpu = of_get_child_by_name(np, "cpu");
 		platform = of_get_child_by_name(np, "platform");
 		codec = of_get_child_by_name(np, "codec");
 
 		if (!cpu) {
-			dev_err(dev, "Can't find cpu DT node\n");
+			dev_err(dev, "%s: Can't find cpu DT node\n", link->name);
 			ret = -EINVAL;
 			goto err;
 		}
@@ -66,7 +72,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 		ret = of_parse_phandle_with_args(cpu, "sound-dai",
 					"#sound-dai-cells", 0, &args);
 		if (ret) {
-			dev_err(card->dev, "error getting cpu phandle\n");
+			dev_err(card->dev, "%s: error getting cpu phandle\n", link->name);
 			goto err;
 		}
 		link->cpus->of_node = args.np;
@@ -74,7 +80,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 
 		ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name);
 		if (ret) {
-			dev_err(card->dev, "error getting cpu dai name\n");
+			dev_err(card->dev, "%s: error getting cpu dai name\n", link->name);
 			goto err;
 		}
 
@@ -83,14 +89,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 					"sound-dai",
 					0);
 			if (!link->platforms->of_node) {
-				dev_err(card->dev, "platform dai not found\n");
+				dev_err(card->dev, "%s: platform dai not found\n", link->name);
 				ret = -EINVAL;
 				goto err;
 			}
 
 			ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
 			if (ret < 0) {
-				dev_err(card->dev, "codec dai not found\n");
+				dev_err(card->dev, "%s: codec dai not found\n", link->name);
 				goto err;
 			}
 			link->no_pcm = 1;
@@ -110,12 +116,6 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 		}
 
 		link->ignore_suspend = 1;
-		ret = of_property_read_string(np, "link-name", &link->name);
-		if (ret) {
-			dev_err(card->dev, "error getting codec dai_link name\n");
-			goto err;
-		}
-
 		link->nonatomic = 1;
 		link->dpcm_playback = 1;
 		link->dpcm_capture = 1;
-- 
cgit v1.2.3-59-g8ed1b


From 8fcd2d914e786033e589b4eb8cb62d37d0fa9701 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 5 Sep 2019 14:01:15 +0200
Subject: ASoC: meson: axg-frddr: expose all 8 outputs

The FRDDR component, as it, has a maximum of 8 outputs. Depending on
the SoC, these may not all be connected.

Instead of decribing only the connected outputs of each SoC, describe
them all and let ASoC routing do the rest.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-frddr.c | 32 ++++++++++++++++++++++++--------
 1 file changed, 24 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
index 2b8807737b2b..0968e8375000 100644
--- a/sound/soc/meson/axg-frddr.c
+++ b/sound/soc/meson/axg-frddr.c
@@ -104,7 +104,7 @@ static struct snd_soc_dai_driver axg_frddr_dai_drv = {
 };
 
 static const char * const axg_frddr_sel_texts[] = {
-	"OUT 0", "OUT 1", "OUT 2", "OUT 3"
+	"OUT 0", "OUT 1", "OUT 2", "OUT 3", "OUT 4", "OUT 5", "OUT 6", "OUT 7",
 };
 
 static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT,
@@ -120,6 +120,10 @@ static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 6", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 7", NULL, 0, SND_SOC_NOPM, 0, 0),
 };
 
 static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = {
@@ -128,6 +132,10 @@ static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = {
 	{ "OUT 1", "OUT 1",  "SINK SEL" },
 	{ "OUT 2", "OUT 2",  "SINK SEL" },
 	{ "OUT 3", "OUT 3",  "SINK SEL" },
+	{ "OUT 4", "OUT 4",  "SINK SEL" },
+	{ "OUT 5", "OUT 5",  "SINK SEL" },
+	{ "OUT 6", "OUT 6",  "SINK SEL" },
+	{ "OUT 7", "OUT 7",  "SINK SEL" },
 };
 
 static const struct snd_soc_component_driver axg_frddr_component_drv = {
@@ -162,16 +170,12 @@ static struct snd_soc_dai_driver g12a_frddr_dai_drv = {
 	.pcm_new	= axg_frddr_pcm_new,
 };
 
-static const char * const g12a_frddr_sel_texts[] = {
-	"OUT 0", "OUT 1", "OUT 2", "OUT 3", "OUT 4",
-};
-
 static SOC_ENUM_SINGLE_DECL(g12a_frddr_sel1_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT,
-			    g12a_frddr_sel_texts);
+			    axg_frddr_sel_texts);
 static SOC_ENUM_SINGLE_DECL(g12a_frddr_sel2_enum, FIFO_CTRL0, CTRL0_SEL2_SHIFT,
-			    g12a_frddr_sel_texts);
+			    axg_frddr_sel_texts);
 static SOC_ENUM_SINGLE_DECL(g12a_frddr_sel3_enum, FIFO_CTRL0, CTRL0_SEL3_SHIFT,
-			    g12a_frddr_sel_texts);
+			    axg_frddr_sel_texts);
 
 static const struct snd_kcontrol_new g12a_frddr_out1_demux =
 	SOC_DAPM_ENUM("Output Src 1", g12a_frddr_sel1_enum);
@@ -211,6 +215,9 @@ static const struct snd_soc_dapm_widget g12a_frddr_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("OUT 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 6", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 7", NULL, 0, SND_SOC_NOPM, 0, 0),
 };
 
 static const struct snd_soc_dapm_route g12a_frddr_dapm_routes[] = {
@@ -228,16 +235,25 @@ static const struct snd_soc_dapm_route g12a_frddr_dapm_routes[] = {
 	{ "OUT 2", "OUT 2", "SINK 1 SEL" },
 	{ "OUT 3", "OUT 3", "SINK 1 SEL" },
 	{ "OUT 4", "OUT 4", "SINK 1 SEL" },
+	{ "OUT 5", "OUT 5", "SINK 1 SEL" },
+	{ "OUT 6", "OUT 6", "SINK 1 SEL" },
+	{ "OUT 7", "OUT 7", "SINK 1 SEL" },
 	{ "OUT 0", "OUT 0", "SINK 2 SEL" },
 	{ "OUT 1", "OUT 1", "SINK 2 SEL" },
 	{ "OUT 2", "OUT 2", "SINK 2 SEL" },
 	{ "OUT 3", "OUT 3", "SINK 2 SEL" },
 	{ "OUT 4", "OUT 4", "SINK 2 SEL" },
+	{ "OUT 5", "OUT 5", "SINK 2 SEL" },
+	{ "OUT 6", "OUT 6", "SINK 2 SEL" },
+	{ "OUT 7", "OUT 7", "SINK 2 SEL" },
 	{ "OUT 0", "OUT 0", "SINK 3 SEL" },
 	{ "OUT 1", "OUT 1", "SINK 3 SEL" },
 	{ "OUT 2", "OUT 2", "SINK 3 SEL" },
 	{ "OUT 3", "OUT 3", "SINK 3 SEL" },
 	{ "OUT 4", "OUT 4", "SINK 3 SEL" },
+	{ "OUT 5", "OUT 5", "SINK 3 SEL" },
+	{ "OUT 6", "OUT 6", "SINK 3 SEL" },
+	{ "OUT 7", "OUT 7", "SINK 3 SEL" },
 };
 
 static const struct snd_soc_component_driver g12a_frddr_component_drv = {
-- 
cgit v1.2.3-59-g8ed1b


From 6beced211c22dd8c3e546c956512fddd8e09884f Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 5 Sep 2019 14:01:16 +0200
Subject: ASoC: meson: axg-toddr: expose all 8 inputs

The TODDR component, as it, has a maximum of 8 input. Depending on
the SoC, these may not all be connected or some input components may
not be supported

Instead of decribing only the connected inputs, describe them all
and let ASoC routing do the rest.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-toddr.c | 15 +++++++--------
 1 file changed, 7 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
index 4f63e434fad4..2e9a2e5862ce 100644
--- a/sound/soc/meson/axg-toddr.c
+++ b/sound/soc/meson/axg-toddr.c
@@ -142,16 +142,11 @@ static struct snd_soc_dai_driver axg_toddr_dai_drv = {
 };
 
 static const char * const axg_toddr_sel_texts[] = {
-	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6"
+	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5", "IN 6", "IN 7"
 };
 
-static const unsigned int axg_toddr_sel_values[] = {
-	0, 1, 2, 3, 4, 6
-};
-
-static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0,
-				  CTRL0_SEL_SHIFT, CTRL0_SEL_MASK,
-				  axg_toddr_sel_texts, axg_toddr_sel_values);
+static SOC_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT,
+			    axg_toddr_sel_texts);
 
 static const struct snd_kcontrol_new axg_toddr_in_mux =
 	SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum);
@@ -163,7 +158,9 @@ static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 7", NULL, 0, SND_SOC_NOPM, 0, 0),
 };
 
 static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = {
@@ -173,7 +170,9 @@ static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = {
 	{ "SRC SEL", "IN 2", "IN 2" },
 	{ "SRC SEL", "IN 3", "IN 3" },
 	{ "SRC SEL", "IN 4", "IN 4" },
+	{ "SRC SEL", "IN 5", "IN 5" },
 	{ "SRC SEL", "IN 6", "IN 6" },
+	{ "SRC SEL", "IN 7", "IN 7" },
 };
 
 static const struct snd_soc_component_driver axg_toddr_component_drv = {
-- 
cgit v1.2.3-59-g8ed1b


From cde9f12e5f0d288d516c810f95c2ee4c171e0245 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 5 Sep 2019 14:01:17 +0200
Subject: ASoC: meson: tdmin: expose all 16 inputs

The TDMIN component, as it, has a maximum of 16 input. Depending on
the SoC, these may not all be connected.

Instead of decribing only the connected inputs of each SoC, describe
them all and let ASoC routing do the rest.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-6-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-tdmin.c | 47 ++++++++++++++++++++++++++++++++-------------
 1 file changed, 34 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index cb87f17f3e95..973d4c02ef8d 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -43,7 +43,8 @@ static const struct regmap_config axg_tdmin_regmap_cfg = {
 };
 
 static const char * const axg_tdmin_sel_texts[] = {
-	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5",
+	"IN 0", "IN 1", "IN 2",  "IN 3",  "IN 4",  "IN 5",  "IN 6",  "IN 7",
+	"IN 8", "IN 9", "IN 10", "IN 11", "IN 12", "IN 13", "IN 14", "IN 15",
 };
 
 /* Change to special mux control to reset dapm */
@@ -164,12 +165,22 @@ static int axg_tdmin_prepare(struct regmap *map,
 }
 
 static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = {
-	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 0",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 3",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 4",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 5",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 6",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 7",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 8",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 9",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 10", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 11", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 12", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 13", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 14", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 15", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux),
 	SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0,
 			   axg_tdm_formatter_event,
@@ -178,12 +189,22 @@ static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = {
-	{ "SRC SEL", "IN 0", "IN 0" },
-	{ "SRC SEL", "IN 1", "IN 1" },
-	{ "SRC SEL", "IN 2", "IN 2" },
-	{ "SRC SEL", "IN 3", "IN 3" },
-	{ "SRC SEL", "IN 4", "IN 4" },
-	{ "SRC SEL", "IN 5", "IN 5" },
+	{ "SRC SEL", "IN 0",  "IN 0" },
+	{ "SRC SEL", "IN 1",  "IN 1" },
+	{ "SRC SEL", "IN 2",  "IN 2" },
+	{ "SRC SEL", "IN 3",  "IN 3" },
+	{ "SRC SEL", "IN 4",  "IN 4" },
+	{ "SRC SEL", "IN 5",  "IN 5" },
+	{ "SRC SEL", "IN 6",  "IN 6" },
+	{ "SRC SEL", "IN 7",  "IN 7" },
+	{ "SRC SEL", "IN 8",  "IN 8" },
+	{ "SRC SEL", "IN 9",  "IN 9" },
+	{ "SRC SEL", "IN 10", "IN 10" },
+	{ "SRC SEL", "IN 11", "IN 11" },
+	{ "SRC SEL", "IN 12", "IN 12" },
+	{ "SRC SEL", "IN 13", "IN 13" },
+	{ "SRC SEL", "IN 14", "IN 14" },
+	{ "SRC SEL", "IN 15", "IN 15" },
 	{ "DEC", NULL, "SRC SEL" },
 	{ "OUT", NULL, "DEC" },
 };
-- 
cgit v1.2.3-59-g8ed1b


From 52dd80d8f7386483bc60b2b7470e47a2e6f61d7c Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 5 Sep 2019 14:01:18 +0200
Subject: ASoC: meson: axg-frddr: add sm1 support

On sm1, the output routing bits have moved to CTRL2 register

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-7-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-fifo.c  |  2 +-
 sound/soc/meson/axg-fifo.h  |  1 +
 sound/soc/meson/axg-frddr.c | 73 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 75 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 80a3dde35b5c..5a3749938900 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -306,7 +306,7 @@ static const struct regmap_config axg_fifo_regmap_cfg = {
 	.reg_bits	= 32,
 	.val_bits	= 32,
 	.reg_stride	= 4,
-	.max_register	= FIFO_INIT_ADDR,
+	.max_register	= FIFO_CTRL2,
 };
 
 int axg_fifo_probe(struct platform_device *pdev)
diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h
index 5caf81241dfe..bb1e2ce50256 100644
--- a/sound/soc/meson/axg-fifo.h
+++ b/sound/soc/meson/axg-fifo.h
@@ -61,6 +61,7 @@ struct snd_soc_pcm_runtime;
 #define  STATUS1_INT_STS(x)		((x) << 0)
 #define FIFO_STATUS2			0x18
 #define FIFO_INIT_ADDR			0x24
+#define FIFO_CTRL2			0x28
 
 struct axg_fifo {
 	struct regmap *map;
diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
index 0968e8375000..6ab111c31b28 100644
--- a/sound/soc/meson/axg-frddr.c
+++ b/sound/soc/meson/axg-frddr.c
@@ -23,6 +23,12 @@
 #define CTRL0_SEL3_SHIFT		8
 #define CTRL0_SEL3_EN_SHIFT		11
 #define CTRL1_FRDDR_FORCE_FINISH	BIT(12)
+#define CTRL2_SEL1_SHIFT		0
+#define CTRL2_SEL1_EN_SHIFT		4
+#define CTRL2_SEL2_SHIFT		8
+#define CTRL2_SEL2_EN_SHIFT		12
+#define CTRL2_SEL3_SHIFT		16
+#define CTRL2_SEL3_EN_SHIFT		20
 
 static int g12a_frddr_dai_prepare(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
@@ -269,6 +275,70 @@ static const struct axg_fifo_match_data g12a_frddr_match_data = {
 	.dai_drv	= &g12a_frddr_dai_drv
 };
 
+/* On SM1, the output selection in on CTRL2 */
+static const struct snd_kcontrol_new sm1_frddr_out1_enable =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", FIFO_CTRL2,
+				    CTRL2_SEL1_EN_SHIFT, 1, 0);
+static const struct snd_kcontrol_new sm1_frddr_out2_enable =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", FIFO_CTRL2,
+				    CTRL2_SEL2_EN_SHIFT, 1, 0);
+static const struct snd_kcontrol_new sm1_frddr_out3_enable =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", FIFO_CTRL2,
+				    CTRL2_SEL3_EN_SHIFT, 1, 0);
+
+static SOC_ENUM_SINGLE_DECL(sm1_frddr_sel1_enum, FIFO_CTRL2, CTRL2_SEL1_SHIFT,
+			    axg_frddr_sel_texts);
+static SOC_ENUM_SINGLE_DECL(sm1_frddr_sel2_enum, FIFO_CTRL2, CTRL2_SEL2_SHIFT,
+			    axg_frddr_sel_texts);
+static SOC_ENUM_SINGLE_DECL(sm1_frddr_sel3_enum, FIFO_CTRL2, CTRL2_SEL3_SHIFT,
+			    axg_frddr_sel_texts);
+
+static const struct snd_kcontrol_new sm1_frddr_out1_demux =
+	SOC_DAPM_ENUM("Output Src 1", sm1_frddr_sel1_enum);
+static const struct snd_kcontrol_new sm1_frddr_out2_demux =
+	SOC_DAPM_ENUM("Output Src 2", sm1_frddr_sel2_enum);
+static const struct snd_kcontrol_new sm1_frddr_out3_demux =
+	SOC_DAPM_ENUM("Output Src 3", sm1_frddr_sel3_enum);
+
+static const struct snd_soc_dapm_widget sm1_frddr_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_OUT("SRC 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SRC 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SRC 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_SWITCH("SRC 1 EN", SND_SOC_NOPM, 0, 0,
+			    &sm1_frddr_out1_enable),
+	SND_SOC_DAPM_SWITCH("SRC 2 EN", SND_SOC_NOPM, 0, 0,
+			    &sm1_frddr_out2_enable),
+	SND_SOC_DAPM_SWITCH("SRC 3 EN", SND_SOC_NOPM, 0, 0,
+			    &sm1_frddr_out3_enable),
+	SND_SOC_DAPM_DEMUX("SINK 1 SEL", SND_SOC_NOPM, 0, 0,
+			   &sm1_frddr_out1_demux),
+	SND_SOC_DAPM_DEMUX("SINK 2 SEL", SND_SOC_NOPM, 0, 0,
+			   &sm1_frddr_out2_demux),
+	SND_SOC_DAPM_DEMUX("SINK 3 SEL", SND_SOC_NOPM, 0, 0,
+			   &sm1_frddr_out3_demux),
+	SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 6", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("OUT 7", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_component_driver sm1_frddr_component_drv = {
+	.dapm_widgets		= sm1_frddr_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(sm1_frddr_dapm_widgets),
+	.dapm_routes		= g12a_frddr_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(g12a_frddr_dapm_routes),
+	.ops			= &g12a_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data sm1_frddr_match_data = {
+	.component_drv	= &sm1_frddr_component_drv,
+	.dai_drv	= &g12a_frddr_dai_drv
+};
+
 static const struct of_device_id axg_frddr_of_match[] = {
 	{
 		.compatible = "amlogic,axg-frddr",
@@ -276,6 +346,9 @@ static const struct of_device_id axg_frddr_of_match[] = {
 	}, {
 		.compatible = "amlogic,g12a-frddr",
 		.data = &g12a_frddr_match_data,
+	}, {
+		.compatible = "amlogic,sm1-frddr",
+		.data = &sm1_frddr_match_data,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_frddr_of_match);
-- 
cgit v1.2.3-59-g8ed1b


From 5ac825c3d85e6c1cb8e43d67d8cb95a2a1e2bc60 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 5 Sep 2019 14:01:19 +0200
Subject: ASoC: meson: axg-toddr: add sm1 support

On sm1, the maximum number TODDR inputs is extended to 16.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-8-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-toddr.c | 68 +++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 68 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
index 2e9a2e5862ce..c8ea2145f576 100644
--- a/sound/soc/meson/axg-toddr.c
+++ b/sound/soc/meson/axg-toddr.c
@@ -25,6 +25,7 @@
 #define CTRL0_TODDR_LSB_POS_MASK	GENMASK(7, 3)
 #define CTRL0_TODDR_LSB_POS(x)		((x) << 3)
 #define CTRL1_TODDR_FORCE_FINISH	BIT(25)
+#define CTRL1_SEL_SHIFT			28
 
 #define TODDR_MSB_POS	31
 
@@ -221,6 +222,70 @@ static const struct axg_fifo_match_data g12a_toddr_match_data = {
 	.dai_drv	= &g12a_toddr_dai_drv
 };
 
+static const char * const sm1_toddr_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2",  "IN 3",  "IN 4",  "IN 5",  "IN 6",  "IN 7",
+	"IN 8", "IN 9", "IN 10", "IN 11", "IN 12", "IN 13", "IN 14", "IN 15"
+};
+
+static SOC_ENUM_SINGLE_DECL(sm1_toddr_sel_enum, FIFO_CTRL1, CTRL1_SEL_SHIFT,
+			    sm1_toddr_sel_texts);
+
+static const struct snd_kcontrol_new sm1_toddr_in_mux =
+	SOC_DAPM_ENUM("Input Source", sm1_toddr_sel_enum);
+
+static const struct snd_soc_dapm_widget sm1_toddr_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &sm1_toddr_in_mux),
+	SND_SOC_DAPM_AIF_IN("IN 0",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 3",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 4",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 5",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 6",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 7",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 8",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 9",  NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 10", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 11", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 12", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 13", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 14", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 15", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route sm1_toddr_dapm_routes[] = {
+	{ "Capture", NULL, "SRC SEL" },
+	{ "SRC SEL", "IN 0",  "IN 0" },
+	{ "SRC SEL", "IN 1",  "IN 1" },
+	{ "SRC SEL", "IN 2",  "IN 2" },
+	{ "SRC SEL", "IN 3",  "IN 3" },
+	{ "SRC SEL", "IN 4",  "IN 4" },
+	{ "SRC SEL", "IN 5",  "IN 5" },
+	{ "SRC SEL", "IN 6",  "IN 6" },
+	{ "SRC SEL", "IN 7",  "IN 7" },
+	{ "SRC SEL", "IN 8",  "IN 8" },
+	{ "SRC SEL", "IN 9",  "IN 9" },
+	{ "SRC SEL", "IN 10", "IN 10" },
+	{ "SRC SEL", "IN 11", "IN 11" },
+	{ "SRC SEL", "IN 12", "IN 12" },
+	{ "SRC SEL", "IN 13", "IN 13" },
+	{ "SRC SEL", "IN 14", "IN 14" },
+	{ "SRC SEL", "IN 15", "IN 15" },
+};
+
+static const struct snd_soc_component_driver sm1_toddr_component_drv = {
+	.dapm_widgets		= sm1_toddr_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(sm1_toddr_dapm_widgets),
+	.dapm_routes		= sm1_toddr_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(sm1_toddr_dapm_routes),
+	.ops			= &g12a_fifo_pcm_ops
+};
+
+static const struct axg_fifo_match_data sm1_toddr_match_data = {
+	.component_drv	= &sm1_toddr_component_drv,
+	.dai_drv	= &g12a_toddr_dai_drv
+};
+
 static const struct of_device_id axg_toddr_of_match[] = {
 	{
 		.compatible = "amlogic,axg-toddr",
@@ -228,6 +293,9 @@ static const struct of_device_id axg_toddr_of_match[] = {
 	}, {
 		.compatible = "amlogic,g12a-toddr",
 		.data = &g12a_toddr_match_data,
+	}, {
+		.compatible = "amlogic,sm1-toddr",
+		.data = &sm1_toddr_match_data,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_toddr_of_match);
-- 
cgit v1.2.3-59-g8ed1b


From 329299d6652fa287a0551ae412cc0227d539a0fe Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 5 Sep 2019 14:01:20 +0200
Subject: ASoC: meson: tdmout: add sm1 support

On the sm1, the TDMOUT number of input is extended and the
the gain enable bit moved to accommodate this extension

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-9-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/meson/axg-tdmout.c | 103 +++++++++++++++++++++++++++++++++++--------
 1 file changed, 84 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 86537fc0ecb5..418ec314b37d 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -24,6 +24,7 @@
 #define TDMOUT_CTRL1			0x04
 #define  TDMOUT_CTRL1_TYPE_MASK		GENMASK(6, 4)
 #define  TDMOUT_CTRL1_TYPE(x)		((x) << 4)
+#define  SM1_TDMOUT_CTRL1_GAIN_EN	7
 #define  TDMOUT_CTRL1_MSB_POS_MASK	GENMASK(12, 8)
 #define  TDMOUT_CTRL1_MSB_POS(x)	((x) << 8)
 #define  TDMOUT_CTRL1_SEL_SHIFT		24
@@ -51,25 +52,6 @@ static const struct regmap_config axg_tdmout_regmap_cfg = {
 	.max_register	= TDMOUT_MASK_VAL,
 };
 
-static const struct snd_kcontrol_new axg_tdmout_controls[] = {
-	SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0,  0,  8, 255, 0),
-	SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0),
-	SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1,  0,  8, 255, 0),
-	SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0),
-	SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1,
-		   TDMOUT_CTRL1_GAIN_EN, 1, 0),
-};
-
-static const char * const tdmout_sel_texts[] = {
-	"IN 0", "IN 1", "IN 2",
-};
-
-static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1,
-			    TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts);
-
-static const struct snd_kcontrol_new axg_tdmout_in_mux =
-	SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum);
-
 static struct snd_soc_dai *
 axg_tdmout_get_be(struct snd_soc_dapm_widget *w)
 {
@@ -197,6 +179,25 @@ static int axg_tdmout_prepare(struct regmap *map,
 	return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0);
 }
 
+static const struct snd_kcontrol_new axg_tdmout_controls[] = {
+	SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0),
+	SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0),
+	SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1,
+		   TDMOUT_CTRL1_GAIN_EN, 1, 0),
+};
+
+static const char * const axg_tdmout_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2",
+};
+
+static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1,
+			    TDMOUT_CTRL1_SEL_SHIFT, axg_tdmout_sel_texts);
+
+static const struct snd_kcontrol_new axg_tdmout_in_mux =
+	SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum);
+
 static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
@@ -252,6 +253,67 @@ static const struct axg_tdm_formatter_driver g12a_tdmout_drv = {
 	},
 };
 
+static const struct snd_kcontrol_new sm1_tdmout_controls[] = {
+	SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0),
+	SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1,  0,  8, 255, 0),
+	SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0),
+	SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1,
+		   SM1_TDMOUT_CTRL1_GAIN_EN, 1, 0),
+};
+
+static const char * const sm1_tdmout_sel_texts[] = {
+	"IN 0", "IN 1", "IN 2", "IN 3", "IN 4",
+};
+
+static SOC_ENUM_SINGLE_DECL(sm1_tdmout_sel_enum, TDMOUT_CTRL1,
+			    TDMOUT_CTRL1_SEL_SHIFT, sm1_tdmout_sel_texts);
+
+static const struct snd_kcontrol_new sm1_tdmout_in_mux =
+	SOC_DAPM_ENUM("Input Source", sm1_tdmout_sel_enum);
+
+static const struct snd_soc_dapm_widget sm1_tdmout_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &sm1_tdmout_in_mux),
+	SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0,
+			   axg_tdm_formatter_event,
+			   (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)),
+	SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route sm1_tdmout_dapm_routes[] = {
+	{ "SRC SEL", "IN 0", "IN 0" },
+	{ "SRC SEL", "IN 1", "IN 1" },
+	{ "SRC SEL", "IN 2", "IN 2" },
+	{ "SRC SEL", "IN 3", "IN 3" },
+	{ "SRC SEL", "IN 4", "IN 4" },
+	{ "ENC", NULL, "SRC SEL" },
+	{ "OUT", NULL, "ENC" },
+};
+
+static const struct snd_soc_component_driver sm1_tdmout_component_drv = {
+	.controls		= sm1_tdmout_controls,
+	.num_controls		= ARRAY_SIZE(sm1_tdmout_controls),
+	.dapm_widgets		= sm1_tdmout_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(sm1_tdmout_dapm_widgets),
+	.dapm_routes		= sm1_tdmout_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(sm1_tdmout_dapm_routes),
+};
+
+static const struct axg_tdm_formatter_driver sm1_tdmout_drv = {
+	.component_drv	= &sm1_tdmout_component_drv,
+	.regmap_cfg	= &axg_tdmout_regmap_cfg,
+	.ops		= &axg_tdmout_ops,
+	.quirks		= &(const struct axg_tdm_formatter_hw) {
+		.invert_sclk = true,
+		.skew_offset = 2,
+	},
+};
+
 static const struct of_device_id axg_tdmout_of_match[] = {
 	{
 		.compatible = "amlogic,axg-tdmout",
@@ -259,6 +321,9 @@ static const struct of_device_id axg_tdmout_of_match[] = {
 	}, {
 		.compatible = "amlogic,g12a-tdmout",
 		.data = &g12a_tdmout_drv,
+	}, {
+		.compatible = "amlogic,sm1-tdmout",
+		.data = &sm1_tdmout_drv,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_tdmout_of_match);
-- 
cgit v1.2.3-59-g8ed1b


From f4df4e4042b045c6ddbaff878a17ae169fe68ba6 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Thu, 5 Sep 2019 14:44:00 +0800
Subject: ASoC: SOF: imx8: Fix COMPILE_TEST error

When do compile test, if SND_SOC_SOF_OF is not set, we get:

sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_request':
imx8.c:(.text+0xb0): undefined reference to `snd_sof_ipc_msgs_rx'
sound/soc/sof/imx/imx8.o: In function `imx8_ipc_msg_data':
imx8.c:(.text+0xf4): undefined reference to `sof_mailbox_read'
sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_reply':
imx8.c:(.text+0x160): undefined reference to `sof_mailbox_read'

Make SND_SOC_SOF_IMX_TOPLEVEL always depends on SND_SOC_SOF_OF

Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190905064400.24800-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/imx/Kconfig | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig
index fd73d8402dbf..5acae75f5750 100644
--- a/sound/soc/sof/imx/Kconfig
+++ b/sound/soc/sof/imx/Kconfig
@@ -2,7 +2,8 @@
 
 config SND_SOC_SOF_IMX_TOPLEVEL
 	bool "SOF support for NXP i.MX audio DSPs"
-	depends on ARM64 && SND_SOC_SOF_OF || COMPILE_TEST
+	depends on ARM64|| COMPILE_TEST
+	depends on SND_SOC_SOF_OF
 	help
           This adds support for Sound Open Firmware for NXP i.MX platforms.
           Say Y if you have such a device.
-- 
cgit v1.2.3-59-g8ed1b


From f953e72c9ca60ef13bb9680af8a5baf92cb93e52 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Thu, 5 Sep 2019 16:48:26 +0100
Subject: ALSA: hda/hdmi: remove redundant assignment to variable pcm_idx

Variable pcm_idx is being initialized with a value that is never read
and is being re-assigned immediately afterwards. The assignment is
redundant and hence can be removed.

Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190905154826.5916-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 83b8b9d27711..bca5de78e9ad 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1440,7 +1440,7 @@ static bool update_eld(struct hda_codec *codec,
 	struct hdmi_spec *spec = codec->spec;
 	bool old_eld_valid = pin_eld->eld_valid;
 	bool eld_changed;
-	int pcm_idx = -1;
+	int pcm_idx;
 
 	/* for monitor disconnection, save pcm_idx firstly */
 	pcm_idx = per_pin->pcm_idx;
-- 
cgit v1.2.3-59-g8ed1b


From 00066e9733f629e536f6b7957de2ce11a85fe15a Mon Sep 17 00:00:00 2001
From: Sergey Bostandzhyan <jin@mediatomb.cc>
Date: Fri, 6 Sep 2019 11:33:43 +0200
Subject: Add Acer Aspire Ethos 8951G model quirk

This notebook has 6 built in speakers for 5.1 surround support, however
only two got autodetected and have also not been assigned correctly.

This patch enables all speakers and also fixes muting when headphones are
plugged in.

The speaker layout is as follows:

pin 0x15 Front Left / Front Right
pin 0x18 Front Center / Subwoofer
pin 0x1b Rear Left / Rear Right (Surround)

The quirk will be enabled automatically on this hardware, but can also be
activated manually via the model=aspire-ethos module parameter.

Caveat: pin 0x1b is shared between headphones jack and surround speakers.
When headphones are plugged in, the surround speakers get muted
automatically by the hardware, however all other speakers remain
unmuted. Currently it's not possible to make use of the generic automute
function in the driver, because such shared pins are not supported.

If we would change the pin settings to identify the pin as headphones,
the surround channel and thus the ability to select 5.1 profiles would
get lost.

This quirk solves the above problem by monitoring jack state of 0x1b and
by connecting/disconnecting all remaining speaker pins when something
gets plugged in or unplugged from the headphones jack port.

Signed-off-by: Sergey Bostandzhyan <jin@mediatomb.cc>
Link: https://lore.kernel.org/r/20190906093343.GA7640@xn--80adja5bqm.su
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 71 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 71 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bddb1a28edc3..7a3c5ca21d32 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8341,6 +8341,45 @@ static void alc662_fixup_usi_headset_mic(struct hda_codec *codec,
 	}
 }
 
+static void alc662_aspire_ethos_mute_speakers(struct hda_codec *codec,
+					struct hda_jack_callback *cb)
+{
+	/* surround speakers at 0x1b already get muted automatically when
+	 * headphones are plugged in, but we have to mute/unmute the remaining
+	 * channels manually:
+	 * 0x15 - front left/front right
+	 * 0x18 - front center/ LFE
+	 */
+	if (snd_hda_jack_detect_state(codec, 0x1b) == HDA_JACK_PRESENT) {
+		snd_hda_set_pin_ctl_cache(codec, 0x15, 0);
+		snd_hda_set_pin_ctl_cache(codec, 0x18, 0);
+	} else {
+		snd_hda_set_pin_ctl_cache(codec, 0x15, PIN_OUT);
+		snd_hda_set_pin_ctl_cache(codec, 0x18, PIN_OUT);
+	}
+}
+
+static void alc662_fixup_aspire_ethos_hp(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+    /* Pin 0x1b: shared headphones jack and surround speakers */
+	if (!is_jack_detectable(codec, 0x1b))
+		return;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_jack_detect_enable_callback(codec, 0x1b,
+				alc662_aspire_ethos_mute_speakers);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		/* Make sure to start in a correct state, i.e. if
+		 * headphones have been plugged in before powering up the system
+		 */
+		alc662_aspire_ethos_mute_speakers(codec, NULL);
+		break;
+	}
+}
+
 static struct coef_fw alc668_coefs[] = {
 	WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03,    0x0),
 	WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06,    0x0), WRITE_COEF(0x07, 0x0f80),
@@ -8412,6 +8451,9 @@ enum {
 	ALC662_FIXUP_USI_FUNC,
 	ALC662_FIXUP_USI_HEADSET_MODE,
 	ALC662_FIXUP_LENOVO_MULTI_CODECS,
+	ALC669_FIXUP_ACER_ASPIRE_ETHOS,
+	ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER,
+	ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -8738,6 +8780,33 @@ static const struct hda_fixup alc662_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc233_alc662_fixup_lenovo_dual_codecs,
 	},
+	[ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc662_fixup_aspire_ethos_hp,
+	},
+	[ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER] = {
+		.type = HDA_FIXUP_VERBS,
+		/* subwoofer needs an extra GPIO setting to become audible */
+		.v.verbs = (const struct hda_verb[]) {
+			{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+			{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET
+	},
+	[ALC669_FIXUP_ACER_ASPIRE_ETHOS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x15, 0x92130110 }, /* front speakers */
+			{ 0x18, 0x99130111 }, /* center/subwoofer */
+			{ 0x1b, 0x11130012 }, /* surround plus jack for HP */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -8783,6 +8852,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68),
 	SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON),
 	SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
+	SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS),
 
 #if 0
 	/* Below is a quirk table taken from the old code.
@@ -8876,6 +8946,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
 	{.id = ALC892_FIXUP_ASROCK_MOBO, .name = "asrock-mobo"},
 	{.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"},
 	{.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"},
+	{.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"},
 	{}
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From e6e2fe2b8f7a4abd24170225050597ebc9f1427b Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Fri, 6 Sep 2019 21:55:44 +0900
Subject: ALSA: firewire-tascam: add note for FE-8

TASCAM FE-8 is the rest of model in TASCAM FireWire series. This device
has no functionality to process audio signal and MIDI messages. Instead,
it transfers control messages to host system corresponding to operations
for some faders, buttons and knobs on its surface.

Unlike the other devices in this series, the control messages are
transmitted by asynchronous transactions. Some registers of device are
used for registration of destination address for the transaction. The
transaction includes quadlet-aligned data up to 32 quadlets.

Userspace applications can receive the transaction and parse it for
control message via Linux FireWire subsystem, without any support by
ALSA firewire-tascam driver. Therefore the driver gives no support
for it.

This commit removes placeholder for FE-8 and add some comment for its
functionalities as notes.

$ python2 linux-firewire-utils/src/crpp < ~/git/am-config-rom/tascam/tascam-fe8.img
               ROM header and bus information block
               -----------------------------------------------------------------
400  040f4798  bus_info_length 4, crc_length 15, crc 18328 (should be 14256)
404  31333934  bus_name "1394"
408  20ff7002  irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256)
40c  00022eff  company_id 00022e     |
410  a094dcb7  device_id ffa094dcb7  | EUI-64 00022effa094dcb7

               root directory
               -----------------------------------------------------------------
414  0004bccc  directory_length 4, crc 48332
418  0300022e  vendor
41c  0c0083c0  node capabilities per IEEE 1394
420  8d000006  --> eui-64 leaf at 438
424  d1000001  --> unit directory at 428

               unit directory at 428
               -----------------------------------------------------------------
428  0003eda4  directory_length 3, crc 60836
42c  1200022e  specifier id
430  13800001  version
434  d4000004  --> dependent info directory at 444

               eui-64 leaf at 438
               -----------------------------------------------------------------
438  0002461e  leaf_length 2, crc 17950
43c  00022eff  company_id 00022e     |
440  a094dcb7  device_id ffa094dcb7  | EUI-64 00022effa094dcb7

               dependent info directory at 444
               -----------------------------------------------------------------
444  0002ae47  directory_length 2, crc 44615
448  81000002  --> descriptor leaf at 450
44c  82000006  --> bus dependent info leaf at 464

               descriptor leaf at 450
               -----------------------------------------------------------------
450  0004a79e  leaf_length 4, crc 42910
454  00000000  textual descriptor
458  00000000  minimal ASCII
45c  54415343  "TASC"
460  414d0000  "AM"

               bus dependent info leaf at 464
               -----------------------------------------------------------------
464  0004a7d8  leaf_length 4, crc 42968
468  00000000
46c  00000000
470  46452d38
474  00000000

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190906125544.13800-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/tascam/tascam.c |  4 +++-
 sound/firewire/tascam/tascam.h | 20 ++++++++++++++++++++
 2 files changed, 23 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index 231052db5680..addc464503bc 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -39,6 +39,9 @@ static const struct snd_tscm_spec model_specs[] = {
 		.midi_capture_ports = 2,
 		.midi_playback_ports = 4,
 	},
+	// This kernel module doesn't support FE-8 because the most of features
+	// can be implemented in userspace without any specific support of this
+	// module.
 };
 
 static int identify_model(struct snd_tscm *tscm)
@@ -214,7 +217,6 @@ static const struct ieee1394_device_id snd_tscm_id_table[] = {
 		.vendor_id = 0x00022e,
 		.specifier_id = 0x00022e,
 	},
-	/* FE-08 requires reverse-engineering because it just has faders. */
 	{}
 };
 MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
index 64a2e4d2bbfe..15bd335fa07f 100644
--- a/sound/firewire/tascam/tascam.h
+++ b/sound/firewire/tascam/tascam.h
@@ -129,6 +129,26 @@ struct snd_tscm {
 
 #define TSCM_OFFSET_MIDI_RX_QUAD	0x4000
 
+// Although FE-8 supports the above registers, it has no I/O interfaces for
+// audio samples and music messages. Otherwise it supports another notification
+// for status and control message as well as LED brightening. The message
+// consists of quadlet-aligned data up to 32 quadlets. The first byte of message
+// is fixed to 0x40. The second byte is between 0x00 to 0x1f and represent each
+// control:
+//   fader:	0x00-0x07
+//   button:	0x0d, 0x0e
+//   knob:	0x14-0x1b
+//   sensing:	0x0b
+//
+// The rest two bytes represent state of the controls; e.g. current value for
+// fader and knob, bitmasks for button and sensing.
+// Just after turning on, 32 quadlets messages with 0x00-0x1f are immediately
+// sent in one transaction. After, several quadlets are sent in one transaction.
+//
+// TSCM_OFFSET_FE8_CTL_TX_ON		0x0310
+// TSCM_OFFSET_FE8_CTL_TX_ADDR_HI	0x0314
+// TSCM_OFFSET_FE8_CTL_TX_ADDR_LO	0x0318
+
 enum snd_tscm_clock {
 	SND_TSCM_CLOCK_INTERNAL = 0,
 	SND_TSCM_CLOCK_WORD	= 1,
-- 
cgit v1.2.3-59-g8ed1b


From 8d0d5c3fa16f050024ff13df106b70466aacbf3b Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Fri, 6 Sep 2019 22:14:14 +0900
Subject: ALSA: firewire-lib: remove WARN_ON() at destruction of AMDTP domain

The destructor of AMDTP domain has WARN_ON() for the list of associated
AMDTP stream. Although this reminds a case that developers forget to
program consumer drivers to stop AMDTP domain, it hits when AMDTP domain
is not initialized yet. This occurs when initialization of sound card
fails as well and it's superfluous.

This commit removes the WARN_ON. Although the API to AMDTP domain does
nothing, it's left for future usage.

Fixes: 3ec3d7a3ff106 ("ALSA: firewire-lib: add AMDTP domain structure to handle several isoc contexts")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190906131414.15370-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 88270257e896..e50e28f77e74 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -1153,7 +1153,8 @@ EXPORT_SYMBOL_GPL(amdtp_domain_init);
  */
 void amdtp_domain_destroy(struct amdtp_domain *d)
 {
-	WARN_ON(!list_empty(&d->streams));
+	// At present nothing to do.
+	return;
 }
 EXPORT_SYMBOL_GPL(amdtp_domain_destroy);
 
-- 
cgit v1.2.3-59-g8ed1b


From 789492f0c86505e63369907bcb1afdf52dec9366 Mon Sep 17 00:00:00 2001
From: Tim Blechmann <tim@klingt.org>
Date: Fri, 6 Sep 2019 16:21:19 +0800
Subject: ALSA: lx6464es - add support for LX6464ESe pci express variant

The pci express variant of the digigram lx6464es card has a different
device ID, but works without changes to the driver.
Thanks to Nikolas Slottke for reporting and testing.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Link: https://lore.kernel.org/r/20190906082119.40971-1-tim@klingt.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/linux/pci_ids.h       | 2 ++
 sound/pci/lx6464es/lx6464es.c | 8 ++++++++
 2 files changed, 10 insertions(+)

(limited to 'sound')

diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h
index c842735a4f45..901c5b2f76de 100644
--- a/include/linux/pci_ids.h
+++ b/include/linux/pci_ids.h
@@ -1951,6 +1951,8 @@
 #define PCI_VENDOR_ID_DIGIGRAM		0x1369
 #define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM	0xc001
 #define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM	0xc002
+#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_SERIAL_SUBSYSTEM		0xc021
+#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_CAE_SERIAL_SUBSYSTEM	0xc022
 
 #define PCI_VENDOR_ID_KAWASAKI		0x136b
 #define PCI_DEVICE_ID_MCHIP_KL5A72002	0xff01
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index 583ca7384d83..fe10714380f2 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -49,6 +49,14 @@ static const struct pci_device_id snd_lx6464es_ids[] = {
 			 PCI_VENDOR_ID_DIGIGRAM,
 			 PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM),
 	},			/* LX6464ES-CAE */
+	{ PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES,
+			 PCI_VENDOR_ID_DIGIGRAM,
+			 PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_SERIAL_SUBSYSTEM),
+	},			/* LX6464ESe */
+	{ PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES,
+			 PCI_VENDOR_ID_DIGIGRAM,
+			 PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_CAE_SERIAL_SUBSYSTEM),
+	},			/* LX6464ESe-CAE */
 	{ 0, },
 };
 
-- 
cgit v1.2.3-59-g8ed1b


From 2ec42f3147e1610716f184b02e65d7f493eed925 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 6 Sep 2019 08:55:24 +0300
Subject: ASoC: dmaengine: Make the pcm->name equal to pcm->id if the name is
 not set

Some tools use the snd_pcm_info_get_name() to try to identify PCMs or for
other purposes.

Currently it is left empty with the dmaengine-pcm, in this case copy the
pcm->id string as pcm->name.

For example IGT is using this to find the HDMI PCM for testing audio on it.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reported-by: Arthur She <arthur.she@linaro.org>
Link: https://lore.kernel.org/r/20190906055524.7393-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 748f5f641002..d93db2c2b527 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -306,6 +306,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
 
 		if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i]))
 			pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE;
+
+		if (rtd->pcm->streams[i].pcm->name[0] == '\0') {
+			strncpy(rtd->pcm->streams[i].pcm->name,
+				rtd->pcm->streams[i].pcm->id,
+				sizeof(rtd->pcm->streams[i].pcm->name));
+		}
 	}
 
 	return 0;
-- 
cgit v1.2.3-59-g8ed1b


From 73681f4f1426847b421649c6aa6c2dc303acc7c8 Mon Sep 17 00:00:00 2001
From: Janusz Krzysztofik <jmkrzyszt@gmail.com>
Date: Sat, 7 Sep 2019 13:16:50 +0200
Subject: ASoC: ams-delta: Take control over audio mute GPIO pins

Since commit 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused
GPIOs"), on-board audio has appeared muted.  It has been discovered that
believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be
set low for audible sound in handsfree and handset mode respectively.

According to Amstrad E3 wiki, the purpose of both pins hasn't been
clearly identified.  Original Amstrad software used to produce a high
pulse on them when the phone was taken off hook or recall was pressed.
With the current findings, we can assume the pins provide a kind of
audio mute function, separately for handset and handsfree operation
modes.

Commit 2afdb4c41d78 ("ARM: OMAP1: ams-delta: Fix audio permanently
muted") attempted to fix the issue temporarily by hogging the GPIO pin
"hookflash1" renamed to "audio_mute", however the fix occurred
incomplete as it restored audible sound only for handsfree mode.

Stop hogging that pin, rename the pins to "handsfree_mute" and
"handset_mute" respectively and implement appropriate DAPM event
callbacks for "Speaker" and "Earpiece" DAPM widgets.

Fixes: 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs")
Signed-off-by: Janusz Krzysztofik <jmkrzyszt@gmail.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190907111650.15440-1-jmkrzyszt@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 arch/arm/mach-omap1/board-ams-delta.c | 10 ++++++----
 sound/soc/ti/ams-delta.c              | 31 +++++++++++++++++++++++++++++--
 2 files changed, 35 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c
index 36498ea1b2f3..c2affc727967 100644
--- a/arch/arm/mach-omap1/board-ams-delta.c
+++ b/arch/arm/mach-omap1/board-ams-delta.c
@@ -245,8 +245,8 @@ static struct platform_device latch2_gpio_device = {
 #define LATCH2_PIN_SCARD_CMDVCC		11
 #define LATCH2_PIN_MODEM_NRESET		12
 #define LATCH2_PIN_MODEM_CODEC		13
-#define LATCH2_PIN_AUDIO_MUTE		14
-#define LATCH2_PIN_HOOKFLASH		15
+#define LATCH2_PIN_HANDSFREE_MUTE	14
+#define LATCH2_PIN_HANDSET_MUTE		15
 
 static struct regulator_consumer_supply modem_nreset_consumers[] = {
 	REGULATOR_SUPPLY("RESET#", "serial8250.1"),
@@ -475,6 +475,10 @@ static struct gpiod_lookup_table ams_delta_audio_gpio_table = {
 			    "hook_switch", 0),
 		GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_MODEM_CODEC,
 			    "modem_codec", 0),
+		GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSFREE_MUTE,
+			    "handsfree_mute", 0),
+		GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSET_MUTE,
+			    "handset_mute", 0),
 		{ },
 	},
 };
@@ -589,8 +593,6 @@ static int gpiochip_match_by_label(struct gpio_chip *chip, void *data)
 static struct gpiod_hog ams_delta_gpio_hogs[] = {
 	GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_KEYBRD_DATAOUT, "keybrd_dataout",
 		 GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW),
-	GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_AUDIO_MUTE, "audio_mute",
-		 GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW),
 	{},
 };
 
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index dee8fc70a64f..8e2fb81ad05c 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -23,14 +23,31 @@
 #include "omap-mcbsp.h"
 #include "../codecs/cx20442.h"
 
+static struct gpio_desc *handset_mute;
+static struct gpio_desc *handsfree_mute;
+
+static int ams_delta_event_handset(struct snd_soc_dapm_widget *w,
+				   struct snd_kcontrol *k, int event)
+{
+	gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w,
+				     struct snd_kcontrol *k, int event)
+{
+	gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
 /* Board specific DAPM widgets */
 static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
 	/* Handset */
 	SND_SOC_DAPM_MIC("Mouthpiece", NULL),
-	SND_SOC_DAPM_HP("Earpiece", NULL),
+	SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset),
 	/* Handsfree/Speakerphone */
 	SND_SOC_DAPM_MIC("Microphone", NULL),
-	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree),
 };
 
 /* How they are connected to codec pins */
@@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
+	handset_mute = devm_gpiod_get(card->dev, "handset_mute",
+				      GPIOD_OUT_HIGH);
+	if (IS_ERR(handset_mute))
+		return PTR_ERR(handset_mute);
+
+	handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute",
+					GPIOD_OUT_HIGH);
+	if (IS_ERR(handsfree_mute))
+		return PTR_ERR(handsfree_mute);
+
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
-- 
cgit v1.2.3-59-g8ed1b


From 0db0c62c88b84b135bbaf784499a08e536354a43 Mon Sep 17 00:00:00 2001
From: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Date: Sun, 8 Sep 2019 01:36:52 +0900
Subject: ASoC: es8316: fix redundant codes of clock

This patch removes redundant null checks for optional MCLK clock.
And fix DT binding document for changing clock property to optional
from required.

Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907163653.9382-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/everest,es8316.txt   |  3 +++
 sound/soc/codecs/es8316.c                          | 31 +++++++++-------------
 2 files changed, 16 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt
index aefcff9c48a2..1bf03c5f2af4 100644
--- a/Documentation/devicetree/bindings/sound/everest,es8316.txt
+++ b/Documentation/devicetree/bindings/sound/everest,es8316.txt
@@ -6,6 +6,9 @@ Required properties:
 
   - compatible  : should be "everest,es8316"
   - reg : the I2C address of the device for I2C
+
+Optional properties:
+
   - clocks : a list of phandle, should contain entries for clock-names
   - clock-names : should include as follows:
          "mclk" : master clock (MCLK) of the device
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 6248b01ca049..e9fa4981ccef 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -370,11 +370,9 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	if (freq == 0)
 		return 0;
 
-	if (es8316->mclk) {
-		ret = clk_set_rate(es8316->mclk, freq);
-		if (ret)
-			return ret;
-	}
+	ret = clk_set_rate(es8316->mclk, freq);
+	if (ret)
+		return ret;
 
 	/* Limit supported sample rates to ones that can be autodetected
 	 * by the codec running in slave mode.
@@ -709,20 +707,18 @@ static int es8316_probe(struct snd_soc_component *component)
 
 	es8316->component = component;
 
-	es8316->mclk = devm_clk_get(component->dev, "mclk");
-	if (PTR_ERR(es8316->mclk) == -EPROBE_DEFER)
-		return -EPROBE_DEFER;
+	es8316->mclk = devm_clk_get_optional(component->dev, "mclk");
 	if (IS_ERR(es8316->mclk)) {
-		dev_err(component->dev, "clock is invalid, ignored\n");
-		es8316->mclk = NULL;
+		dev_err(component->dev, "unable to get mclk\n");
+		return PTR_ERR(es8316->mclk);
 	}
+	if (!es8316->mclk)
+		dev_warn(component->dev, "assuming static mclk\n");
 
-	if (es8316->mclk) {
-		ret = clk_prepare_enable(es8316->mclk);
-		if (ret) {
-			dev_err(component->dev, "unable to enable clock\n");
-			return ret;
-		}
+	ret = clk_prepare_enable(es8316->mclk);
+	if (ret) {
+		dev_err(component->dev, "unable to enable mclk\n");
+		return ret;
 	}
 
 	/* Reset codec and enable current state machine */
@@ -751,8 +747,7 @@ static void es8316_remove(struct snd_soc_component *component)
 {
 	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
 
-	if (es8316->mclk)
-		clk_disable_unprepare(es8316->mclk);
+	clk_disable_unprepare(es8316->mclk);
 }
 
 static const struct snd_soc_component_driver soc_component_dev_es8316 = {
-- 
cgit v1.2.3-59-g8ed1b


From ebe02a5b9ef05e3b812af3d628cdf6206d9ba610 Mon Sep 17 00:00:00 2001
From: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Date: Sun, 8 Sep 2019 01:36:53 +0900
Subject: ASoC: es8316: support fixed and variable both clock rates

This patch supports some type of machine drivers that set 0 to mclk
when sound device goes to idle state. After applied this patch,
sysclk == 0 means there is no constraint of sound rate and other
values will set constraints which is derived by sysclk setting.

Original code refuses sysclk == 0 setting. But some boards and SoC
(such as RockPro64 and RockChip I2S) has connected SoC MCLK out to
ES8316 MCLK in. In this case, SoC side I2S will choose suitable
frequency of MCLK such as fs * mclk-fs when user starts playing or
capturing.

Bad scenario as follows (mclk-fs = 256):
  - Initialize sysclk by correct value (Ex. 12.288MHz)
    - ES8316 set constraints of PCM rate by sysclk
      48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400),
      24kHz (1/512), 16kHz (1/768), 12kHz (1/1024)
  - Play 48kHz sound, it's acceptable
  - Sysclk is not changed

  - Play 32kHz sound, it's acceptable
  - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256)
    - ES8316 set constraints of PCM rate by sysclk
      32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400),
      16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024)

  - Play 48kHz again, but it's NOT acceptable because constraints
    list does not allow 48kHz

Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907163653.9382-2-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8316.c | 35 ++++++++++++++++++++---------------
 1 file changed, 20 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index e9fa4981ccef..9150e7068467 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -367,8 +367,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 
 	es8316->sysclk = freq;
 
-	if (freq == 0)
+	if (freq == 0) {
+		es8316->sysclk_constraints.list = NULL;
+		es8316->sysclk_constraints.count = 0;
+
 		return 0;
+	}
 
 	ret = clk_set_rate(es8316->mclk, freq);
 	if (ret)
@@ -447,17 +451,10 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
 
-	if (es8316->sysclk == 0) {
-		dev_err(component->dev, "No sysclk provided\n");
-		return -EINVAL;
-	}
-
-	/* The set of sample rates that can be supported depends on the
-	 * MCLK supplied to the CODEC.
-	 */
-	snd_pcm_hw_constraint_list(substream->runtime, 0,
-				   SNDRV_PCM_HW_PARAM_RATE,
-				   &es8316->sysclk_constraints);
+	if (es8316->sysclk_constraints.list)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_RATE,
+					   &es8316->sysclk_constraints);
 
 	return 0;
 }
@@ -469,11 +466,19 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
 	u8 wordlen = 0;
+	int i;
 
-	if (!es8316->sysclk) {
-		dev_err(component->dev, "No MCLK configured\n");
-		return -EINVAL;
+	/* Validate supported sample rates that are autodetected from MCLK */
+	for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) {
+		const unsigned int ratio = supported_mclk_lrck_ratios[i];
+
+		if (es8316->sysclk % ratio != 0)
+			continue;
+		if (es8316->sysclk / ratio == params_rate(params))
+			break;
 	}
+	if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS)
+		return -EINVAL;
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-- 
cgit v1.2.3-59-g8ed1b


From 78b93b04771bdbefe94b84222506ba992f579c98 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Fri, 6 Sep 2019 17:14:04 +0100
Subject: ASoC: codecs: ad193x: make two arrays static const, makes object
 smaller

Don't populate the arrays on the stack but instead make them
static const. Makes the object code smaller by 37 bytes.

Before:
   text	   data	    bss	    dec	    hex	filename
  16253	   7200	      0	  23453	   5b9d	sound/soc/codecs/ad193x.o

After:
   text	   data	    bss	    dec	    hex	filename
  16056	   7360	      0	  23416	   5b78	sound/soc/codecs/ad193x.o

(gcc version 9.2.1, amd64)

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190906161404.1440-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ad193x.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index fb04c9379b71..980e024a5720 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -416,7 +416,7 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = {
 /* codec register values to set after reset */
 static void ad193x_reg_default_init(struct ad193x_priv *ad193x)
 {
-	const struct reg_sequence reg_init[] = {
+	static const struct reg_sequence reg_init[] = {
 		{  0, 0x99 },	/* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */
 		{  1, 0x04 },	/* PLL_CLK_CTRL1: no on-chip Vref */
 		{  2, 0x40 },	/* DAC_CTRL0: TDM mode */
@@ -432,7 +432,7 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x)
 		{ 12, 0x00 },	/* DAC_L4_VOL: no attenuation */
 		{ 13, 0x00 },	/* DAC_R4_VOL: no attenuation */
 	};
-	const struct reg_sequence reg_adc_init[] = {
+	static const struct reg_sequence reg_adc_init[] = {
 		{ 14, 0x03 },	/* ADC_CTRL0: high-pass filter enable */
 		{ 15, 0x43 },	/* ADC_CTRL1: sata delay=1, adc aux mode */
 		{ 16, 0x00 },	/* ADC_CTRL2: reset */
-- 
cgit v1.2.3-59-g8ed1b


From f1879d7b98dc9081cffc7718b644c6c41628cd18 Mon Sep 17 00:00:00 2001
From: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Date: Sun, 8 Sep 2019 02:43:32 +0900
Subject: ASoC: rockchip: ignore 0Hz sysclk

This patch ignores sysclk setting if it is 0Hz.

Some codecs treat 0Hz sysclk as signal of applying no constraints.
This driver does not have such feature but current implementation
outputs 'Failed to set mclk' error message if machine driver sets
0Hz sysclk to this driver.

Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907174332.19586-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_i2s.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 88ebaf6e1880..af2d5a6124c8 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -419,6 +419,9 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
 	struct rk_i2s_dev *i2s = to_info(cpu_dai);
 	int ret;
 
+	if (freq == 0)
+		return 0;
+
 	ret = clk_set_rate(i2s->mclk, freq);
 	if (ret)
 		dev_err(i2s->dev, "Fail to set mclk %d\n", ret);
-- 
cgit v1.2.3-59-g8ed1b


From 2458adb8f92ad4d07ef7ab27c5bafa1d3f4678d6 Mon Sep 17 00:00:00 2001
From: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Date: Sun, 8 Sep 2019 02:45:01 +0900
Subject: SoC: simple-card-utils: set 0Hz to sysclk when shutdown

This patch set 0Hz to sysclk when shutdown the card.

Some codecs set rate constraints that derives from sysclk. This
mechanism works correctly if machine drivers give fixed frequency.

But simple-audio and audio-graph card set variable clock rate if
'mclk-fs' property exists. In this case, rate constraints will go
bad scenario. For example a codec accepts three limited rates
(mclk / 256, mclk / 384, mclk / 512).

Bad scenario as follows (mclk-fs = 256):
   - Initialize sysclk by correct value (Ex. 12.288MHz)
     - Codec set constraints of PCM rate by sysclk
       48kHz (1/256), 32kHz (1/384), 24kHz (1/512)
   - Play 48kHz sound, it's acceptable
   - Sysclk is not changed

   - Play 32kHz sound, it's acceptable
   - Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256)
     - Codec set constraints of PCM rate by sysclk
       32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512)

   - Play 48kHz again, but it's NOT acceptable because constraints
     do not allow 48kHz

So codecs treat 0Hz sysclk as signal of applying no constraints to
avoid this problem.

Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907174501.19833-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card-utils.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 556b1a789629..9b794775df53 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -213,10 +213,17 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup);
 void asoc_simple_shutdown(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
+	if (dai_props->mclk_fs) {
+		snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN);
+		snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT);
+	}
+
 	asoc_simple_clk_disable(dai_props->cpu_dai);
 
 	asoc_simple_clk_disable(dai_props->codec_dai);
-- 
cgit v1.2.3-59-g8ed1b


From 2b70d5776e8b173e3b36a2ef63d94428c6a80e1f Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Sat, 7 Sep 2019 08:37:17 +0100
Subject: ASoC: rt1011: make array pd static const, makes object smaller

Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 100 bytes.

Before:
   text	   data	    bss	    dec	    hex	filename
  51463	  13016	    128	  64607	   fc5f	sound/soc/codecs/rt1011.o

After:
   text	   data	    bss	    dec	    hex	filename
  51299	  13080	    128	  64507	   fbfb	sound/soc/codecs/rt1011.o

(gcc version 9.2.1, amd64)

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907073717.21632-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1011.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index fa34565a3938..a92a0bacd812 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -1519,7 +1519,8 @@ static const struct snd_soc_dapm_route rt1011_dapm_routes[] = {
 
 static int rt1011_get_clk_info(int sclk, int rate)
 {
-	int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
+	int i;
+	static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
 
 	if (sclk <= 0 || rate <= 0)
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From b61b1e35ed06db180968cb5ca0fbf8b8887ccb93 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Sat, 7 Sep 2019 08:41:56 +0100
Subject: ASoC: rt1305: make array pd static const, makes object smaller

Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 93 bytes.

Before:
   text	   data	    bss	    dec	    hex	filename
  38961	   9784	     64	  48809	   bea9	sound/soc/codecs/rt1305.o

After:
   text	   data	    bss	    dec	    hex	filename
  38804	   9848	     64	  48716	   be4c	sound/soc/codecs/rt1305.o

(gcc version 9.2.1, amd64)

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907074156.21907-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1305.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c
index 9909369483f0..e27742abfa76 100644
--- a/sound/soc/codecs/rt1305.c
+++ b/sound/soc/codecs/rt1305.c
@@ -608,7 +608,8 @@ static const struct snd_soc_dapm_route rt1305_dapm_routes[] = {
 
 static int rt1305_get_clk_info(int sclk, int rate)
 {
-	int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
+	int i;
+	static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
 
 	if (sclk <= 0 || rate <= 0)
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From 13ab0d1ab8ae4d1b3b247fd5caec2c10e35c7269 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Sat, 7 Sep 2019 08:46:34 +0100
Subject: ASoC: rt1308: make array pd static const, makes object smaller

Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 82 bytes.

Before:
   text	   data	    bss	    dec	    hex	filename
  26548	   7288	     64	  33900	   846c	sound/soc/codecs/rt1308.o

After:
   text	   data	    bss	    dec	    hex	filename
  26370	   7384	     64	  33818	   841a	sound/soc/codecs/rt1308.o

(gcc version 9.2.1, amd64)

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907074634.22144-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1308.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
index 6f2ee6809dbb..b75931a69a1c 100644
--- a/sound/soc/codecs/rt1308.c
+++ b/sound/soc/codecs/rt1308.c
@@ -436,7 +436,8 @@ static const struct snd_soc_dapm_route rt1308_dapm_routes[] = {
 
 static int rt1308_get_clk_info(int sclk, int rate)
 {
-	int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
+	int i;
+	static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
 
 	if (sclk <= 0 || rate <= 0)
 		return -EINVAL;
-- 
cgit v1.2.3-59-g8ed1b


From 62f07a6b6dbaf0e7196b6911111d4666b5d03518 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:14:46 +0900
Subject: ASoC: soc-core: self contained soc_probe_link_components()

Current soc_probe_link_components() implementation is very half,
thus it is very difficult to read.

	for_each_comp_order(xxx) {
		for_each_card_rtds(xxx) {
=>			ret = soc_probe_link_components(xxx);
			...
		}
	}

This patch does all for_each_xxx() under soc_probe_link_components(),
and makes it to self contained.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874l1sq5mx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 38 +++++++++++++++++++-------------------
 1 file changed, 19 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 05a2aff843aa..04b98e6c7b2a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1137,20 +1137,25 @@ static void soc_remove_link_components(struct snd_soc_card *card,
 	}
 }
 
-static int soc_probe_link_components(struct snd_soc_card *card,
-				     struct snd_soc_pcm_runtime *rtd, int order)
+static int soc_probe_link_components(struct snd_soc_card *card)
 {
 	struct snd_soc_component *component;
+	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_rtdcom_list *rtdcom;
-	int ret;
+	int ret, order;
 
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
+			for_each_rtdcom(rtd, rtdcom) {
+				component = rtdcom->component;
 
-		if (component->driver->probe_order == order) {
-			ret = soc_probe_component(card, component);
-			if (ret < 0)
-				return ret;
+				if (component->driver->probe_order != order)
+					continue;
+
+				ret = soc_probe_component(card, component);
+				if (ret < 0)
+					return ret;
+			}
 		}
 	}
 
@@ -1990,16 +1995,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* probe all components used by DAI links on this card */
-	for_each_comp_order(order) {
-		for_each_card_rtds(card, rtd) {
-			ret = soc_probe_link_components(card, rtd, order);
-			if (ret < 0) {
-				dev_err(card->dev,
-					"ASoC: failed to instantiate card %d\n",
-					ret);
-				goto probe_end;
-			}
-		}
+	ret = soc_probe_link_components(card);
+	if (ret < 0) {
+		dev_err(card->dev,
+			"ASoC: failed to instantiate card %d\n", ret);
+		goto probe_end;
 	}
 
 	/* probe auxiliary components */
-- 
cgit v1.2.3-59-g8ed1b


From b006c0c6ed0dc779e64a8d7a25f5cf316fa3562c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:14:51 +0900
Subject: ASoC: soc-core: self contained soc_remove_link_components()

Current soc_remove_link_components() implementation is very half,
thus it is very difficult to read.

	for_each_comp_order(xxx) {
		for_each_card_rtds(xxx)
=>			soc_remove_link_components(xxx);
	}

This patch does all for_each_xxx() under soc_remove_link_components(),
and makes it to self contained.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736hcq5ms.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 33 +++++++++++++++------------------
 1 file changed, 15 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 04b98e6c7b2a..fbaf4dd597b5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1123,17 +1123,24 @@ static void soc_remove_link_dais(struct snd_soc_card *card,
 	soc_remove_dai(rtd->cpu_dai, order);
 }
 
-static void soc_remove_link_components(struct snd_soc_card *card,
-	struct snd_soc_pcm_runtime *rtd, int order)
+static void soc_remove_link_components(struct snd_soc_card *card)
 {
 	struct snd_soc_component *component;
+	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_rtdcom_list *rtdcom;
+	int order;
 
-	for_each_rtdcom(rtd, rtdcom) {
-		component = rtdcom->component;
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
+			for_each_rtdcom(rtd, rtdcom) {
+				component = rtdcom->component;
+
+				if (component->driver->remove_order != order)
+					continue;
 
-		if (component->driver->remove_order == order)
-			soc_remove_component(component);
+				soc_remove_component(component);
+			}
+		}
 	}
 }
 
@@ -1173,10 +1180,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
 			soc_remove_link_dais(card, rtd, order);
 	}
 
-	for_each_comp_order(order) {
-		for_each_card_rtds(card, rtd)
-			soc_remove_link_components(card, rtd, order);
-	}
+	soc_remove_link_components(card);
 
 	for_each_card_links_safe(card, link, _link) {
 		if (link->dobj.type == SND_SOC_DOBJ_DAI_LINK)
@@ -2394,20 +2398,13 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
 static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
 {
-	struct snd_soc_pcm_runtime *rtd;
-	int order;
-
 	if (card->instantiated) {
 		card->instantiated = false;
 		snd_soc_dapm_shutdown(card);
 		snd_soc_flush_all_delayed_work(card);
 
 		/* remove all components used by DAI links on this card */
-		for_each_comp_order(order) {
-			for_each_card_rtds(card, rtd) {
-				soc_remove_link_components(card, rtd, order);
-			}
-		}
+		soc_remove_link_components(card);
 
 		soc_cleanup_card_resources(card);
 		if (!unregister)
-- 
cgit v1.2.3-59-g8ed1b


From 4ca47d21bf7aa044c703688fa715862183dd6a3c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:14:57 +0900
Subject: ASoC: soc-core: self contained soc_remove_link_dais()

Current soc_remove_link_dais() implementation is very half,
thus it is very difficult to read.

	for_each_comp_order(xxx) {
		for_each_card_rtds(xxx)
=>			soc_remove_link_dais(xxx);
	}

This patch does all for_each_xxx() under soc_remove_link_dais(),
and makes it to self contained.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871rwwq5mm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 29 +++++++++++++++--------------
 1 file changed, 15 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fbaf4dd597b5..0a13f9461303 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1107,20 +1107,26 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 }
 
 static void soc_rtd_free(struct snd_soc_pcm_runtime *rtd); /* remove me */
-static void soc_remove_link_dais(struct snd_soc_card *card,
-		struct snd_soc_pcm_runtime *rtd, int order)
+static void soc_remove_link_dais(struct snd_soc_card *card)
 {
 	int i;
 	struct snd_soc_dai *codec_dai;
+	struct snd_soc_pcm_runtime *rtd;
+	int order;
+
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
 
-	/* finalize rtd device */
-	soc_rtd_free(rtd);
+			/* finalize rtd device */
+			soc_rtd_free(rtd);
 
-	/* remove the CODEC DAI */
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
-		soc_remove_dai(codec_dai, order);
+			/* remove the CODEC DAI */
+			for_each_rtd_codec_dai(rtd, i, codec_dai)
+				soc_remove_dai(codec_dai, order);
 
-	soc_remove_dai(rtd->cpu_dai, order);
+			soc_remove_dai(rtd->cpu_dai, order);
+		}
+	}
 }
 
 static void soc_remove_link_components(struct snd_soc_card *card)
@@ -1171,14 +1177,9 @@ static int soc_probe_link_components(struct snd_soc_card *card)
 
 static void soc_remove_dai_links(struct snd_soc_card *card)
 {
-	int order;
-	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai_link *link, *_link;
 
-	for_each_comp_order(order) {
-		for_each_card_rtds(card, rtd)
-			soc_remove_link_dais(card, rtd, order);
-	}
+	soc_remove_link_dais(card);
 
 	soc_remove_link_components(card);
 
-- 
cgit v1.2.3-59-g8ed1b


From a7d44f78063d642406132c2ed1dc6033d0d03cfe Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:15:06 +0900
Subject: ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai()

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

This patch moves soc_probe_dai() next to soc_remove_dai() which is
paired function.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhjkor1x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 40 ++++++++++++++++++++--------------------
 1 file changed, 20 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0a13f9461303..3ca5a21bfa6a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1106,6 +1106,26 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 	dai->probed = 0;
 }
 
+static int soc_probe_dai(struct snd_soc_dai *dai, int order)
+{
+	int ret;
+
+	if (dai->probed ||
+	    dai->driver->probe_order != order)
+		return 0;
+
+	ret = snd_soc_dai_probe(dai);
+	if (ret < 0) {
+		dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
+			dai->name, ret);
+		return ret;
+	}
+
+	dai->probed = 1;
+
+	return 0;
+}
+
 static void soc_rtd_free(struct snd_soc_pcm_runtime *rtd); /* remove me */
 static void soc_remove_link_dais(struct snd_soc_card *card)
 {
@@ -1411,26 +1431,6 @@ static int soc_rtd_init(struct snd_soc_pcm_runtime *rtd, const char *name)
 	return 0;
 }
 
-static int soc_probe_dai(struct snd_soc_dai *dai, int order)
-{
-	int ret;
-
-	if (dai->probed ||
-	    dai->driver->probe_order != order)
-		return 0;
-
-	ret = snd_soc_dai_probe(dai);
-	if (ret < 0) {
-		dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
-			dai->name, ret);
-		return ret;
-	}
-
-	dai->probed = 1;
-
-	return 0;
-}
-
 static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
 				struct snd_soc_pcm_runtime *rtd)
 {
-- 
cgit v1.2.3-59-g8ed1b


From c4b4698291a1d1c55665ae906779d1b685d5aa66 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:15:12 +0900
Subject: ASoC: soc-core: add new soc_link_init()

Current soc_probe_link_dais() (1) is called under probe_order (2),
and it will initialize dai_link related settings at *Last* turn (3)(B).
It is very complex code.

	static int soc_probe_link_dais(..., order)
	{
(A)		/* probe DAIs here */
		...

(3)		if (order != SND_SOC_COMP_ORDER_LAST)
			return 0;

(B)		/* initialize dai_link related settings */
		...
	}

	static int snd_soc_instantiate_card(...)
	{
		...
(2)		for_each_comp_order(order) {
			for_each_card_rtds(...) {
(1)				ret = soc_probe_link_dais(..., order);
			}
		}
	}

This patch separes soc_probe_link_dais() into "DAI probe" portion (A),
and dai_link settings portion (B).
The later is named as soc_link_init() by this patch.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2z4or1r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 28 ++++++++++++++++++----------
 1 file changed, 18 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3ca5a21bfa6a..1e74ab8e6a25 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1455,19 +1455,13 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
 static int soc_probe_link_dais(struct snd_soc_card *card,
 		struct snd_soc_pcm_runtime *rtd, int order)
 {
-	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_rtdcom_list *rtdcom;
-	struct snd_soc_component *component;
 	struct snd_soc_dai *codec_dai;
-	int i, ret, num;
+	int i, ret;
 
 	dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
 			card->name, rtd->num, order);
 
-	/* set default power off timeout */
-	rtd->pmdown_time = pmdown_time;
-
 	ret = soc_probe_dai(cpu_dai, order);
 	if (ret)
 		return ret;
@@ -1479,9 +1473,20 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 			return ret;
 	}
 
-	/* complete DAI probe during last probe */
-	if (order != SND_SOC_COMP_ORDER_LAST)
-		return 0;
+	return 0;
+}
+
+static int soc_link_init(struct snd_soc_card *card,
+			 struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_rtdcom_list *rtdcom;
+	struct snd_soc_component *component;
+	int ret, num;
+
+	/* set default power off timeout */
+	rtd->pmdown_time = pmdown_time;
 
 	/* do machine specific initialization */
 	if (dai_link->init) {
@@ -2041,6 +2046,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		}
 	}
 
+	for_each_card_rtds(card, rtd)
+		soc_link_init(card, rtd);
+
 	snd_soc_dapm_link_dai_widgets(card);
 	snd_soc_dapm_connect_dai_link_widgets(card);
 
-- 
cgit v1.2.3-59-g8ed1b


From c7e73774f2f4213e75670f393a7b253620e04441 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:15:17 +0900
Subject: ASoC: soc-core: self contained soc_probe_link_dais()

Current soc_probe_link_dais() implementation is very half,
thus it is very difficult to read.

	for_each_comp_order(xxx) {
		for_each_card_rtds(xxx)
=>			soc_probe_link_dais(xxx);
	}

This patch does all for_each_xxx() under soc_probe_link_dais(),
and makes it to self contained.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87woeoor1m.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 50 +++++++++++++++++++++++++-------------------------
 1 file changed, 25 insertions(+), 25 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1e74ab8e6a25..36b86e3c193c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1452,25 +1452,30 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
 	return 0;
 }
 
-static int soc_probe_link_dais(struct snd_soc_card *card,
-		struct snd_soc_pcm_runtime *rtd, int order)
+static int soc_probe_link_dais(struct snd_soc_card *card)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai;
-	int i, ret;
+	struct snd_soc_pcm_runtime *rtd;
+	int i, order, ret;
 
-	dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
-			card->name, rtd->num, order);
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
 
-	ret = soc_probe_dai(cpu_dai, order);
-	if (ret)
-		return ret;
+			dev_dbg(card->dev,
+				"ASoC: probe %s dai link %d late %d\n",
+				card->name, rtd->num, order);
 
-	/* probe the CODEC DAI */
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		ret = soc_probe_dai(codec_dai, order);
-		if (ret)
-			return ret;
+			ret = soc_probe_dai(rtd->cpu_dai, order);
+			if (ret)
+				return ret;
+
+			/* probe the CODEC DAI */
+			for_each_rtd_codec_dai(rtd, i, codec_dai) {
+				ret = soc_probe_dai(codec_dai, order);
+				if (ret)
+					return ret;
+			}
+		}
 	}
 
 	return 0;
@@ -1933,7 +1938,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai_link *dai_link;
 	struct snd_soc_aux_dev *aux;
-	int ret, i, order;
+	int ret, i;
 
 	mutex_lock(&client_mutex);
 	for_each_card_prelinks(card, i, dai_link) {
@@ -2034,16 +2039,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* probe all DAI links on this card */
-	for_each_comp_order(order) {
-		for_each_card_rtds(card, rtd) {
-			ret = soc_probe_link_dais(card, rtd, order);
-			if (ret < 0) {
-				dev_err(card->dev,
-					"ASoC: failed to instantiate card %d\n",
-					ret);
-				goto probe_end;
-			}
-		}
+	ret = soc_probe_link_dais(card);
+	if (ret < 0) {
+		dev_err(card->dev,
+			"ASoC: failed to instantiate card %d\n", ret);
+		goto probe_end;
 	}
 
 	for_each_card_rtds(card, rtd)
-- 
cgit v1.2.3-59-g8ed1b


From bc7c16c226a919a509e4161c905bfe9981a17f02 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:15:23 +0900
Subject: ASoC: soc-core: move soc_probe_link_dais() next to
 soc_remove_link_dais()

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

This patch moves soc_probe_link_dais() next to soc_remove_link_dais()
which is paired function.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v9u8or1g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++--------------------------
 1 file changed, 29 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 36b86e3c193c..20381f69a34b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1149,6 +1149,35 @@ static void soc_remove_link_dais(struct snd_soc_card *card)
 	}
 }
 
+static int soc_probe_link_dais(struct snd_soc_card *card)
+{
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_pcm_runtime *rtd;
+	int i, order, ret;
+
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
+
+			dev_dbg(card->dev,
+				"ASoC: probe %s dai link %d late %d\n",
+				card->name, rtd->num, order);
+
+			ret = soc_probe_dai(rtd->cpu_dai, order);
+			if (ret)
+				return ret;
+
+			/* probe the CODEC DAI */
+			for_each_rtd_codec_dai(rtd, i, codec_dai) {
+				ret = soc_probe_dai(codec_dai, order);
+				if (ret)
+					return ret;
+			}
+		}
+	}
+
+	return 0;
+}
+
 static void soc_remove_link_components(struct snd_soc_card *card)
 {
 	struct snd_soc_component *component;
@@ -1452,35 +1481,6 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
 	return 0;
 }
 
-static int soc_probe_link_dais(struct snd_soc_card *card)
-{
-	struct snd_soc_dai *codec_dai;
-	struct snd_soc_pcm_runtime *rtd;
-	int i, order, ret;
-
-	for_each_comp_order(order) {
-		for_each_card_rtds(card, rtd) {
-
-			dev_dbg(card->dev,
-				"ASoC: probe %s dai link %d late %d\n",
-				card->name, rtd->num, order);
-
-			ret = soc_probe_dai(rtd->cpu_dai, order);
-			if (ret)
-				return ret;
-
-			/* probe the CODEC DAI */
-			for_each_rtd_codec_dai(rtd, i, codec_dai) {
-				ret = soc_probe_dai(codec_dai, order);
-				if (ret)
-					return ret;
-			}
-		}
-	}
-
-	return 0;
-}
-
 static int soc_link_init(struct snd_soc_card *card,
 			 struct snd_soc_pcm_runtime *rtd)
 {
-- 
cgit v1.2.3-59-g8ed1b


From bee886f1ea9d9e58bef8204c543a409c399f8744 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:15:28 +0900
Subject: ASoC: soc-core: self contained soc_bind_aux_dev()

Current soc_bind_aux_dev() implementation is very half,
thus it is very unreadable.

	for_each_card_pre_auxs(xxx) {
=>		ret = soc_bind_aux_dev(xxx);
		...
	}

This patch does all for_each_xxx() under soc_bind_aux_dev(),
and makes it to self contained.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tv9sor1b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 31 +++++++++++++++----------------
 1 file changed, 15 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 20381f69a34b..d7761df03e19 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1559,20 +1559,22 @@ static int soc_link_init(struct snd_soc_card *card,
 	return ret;
 }
 
-static int soc_bind_aux_dev(struct snd_soc_card *card,
-			    struct snd_soc_aux_dev *aux_dev)
+static int soc_bind_aux_dev(struct snd_soc_card *card)
 {
 	struct snd_soc_component *component;
+	struct snd_soc_aux_dev *aux;
+	int i;
 
-	/* codecs, usually analog devices */
-	component = soc_find_component(&aux_dev->dlc);
-	if (!component)
-		return -EPROBE_DEFER;
-
-	component->init = aux_dev->init;
-	/* see for_each_card_auxs */
-	list_add(&component->card_aux_list, &card->aux_comp_list);
+	for_each_card_pre_auxs(card, i, aux) {
+		/* codecs, usually analog devices */
+		component = soc_find_component(&aux->dlc);
+		if (!component)
+			return -EPROBE_DEFER;
 
+		component->init = aux->init;
+		/* see for_each_card_auxs */
+		list_add(&component->card_aux_list, &card->aux_comp_list);
+	}
 	return 0;
 }
 
@@ -1937,7 +1939,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai_link *dai_link;
-	struct snd_soc_aux_dev *aux;
 	int ret, i;
 
 	mutex_lock(&client_mutex);
@@ -1965,11 +1966,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* bind aux_devs too */
-	for_each_card_pre_auxs(card, i, aux) {
-		ret = soc_bind_aux_dev(card, aux);
-		if (ret != 0)
-			goto probe_end;
-	}
+	ret = soc_bind_aux_dev(card);
+	if (ret < 0)
+		goto probe_end;
 
 	/* add predefined DAI links to the list */
 	for_each_card_prelinks(card, i, dai_link) {
-- 
cgit v1.2.3-59-g8ed1b


From 4893a2eb34a9722501915a0b75844aa11956c10d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:15:35 +0900
Subject: ASoC: soc-core: add soc_unbind_aux_dev()

It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.

soc-core.c has soc_bind_aux_dev(), but, there is no its paired
soc_unbind_aux_dev().
This patch adds soc_unbind_aux_dev().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgpcor14.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d7761df03e19..2960070e68bc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1559,6 +1559,12 @@ static int soc_link_init(struct snd_soc_card *card,
 	return ret;
 }
 
+static void soc_unbind_aux_dev(struct snd_soc_component *component)
+{
+	component->init = NULL;
+	list_del(&component->card_aux_list);
+}
+
 static int soc_bind_aux_dev(struct snd_soc_card *card)
 {
 	struct snd_soc_component *component;
@@ -1612,7 +1618,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 			if (comp->driver->remove_order == order) {
 				soc_remove_component(comp);
 				/* remove it from the card's aux_comp_list */
-				list_del(&comp->card_aux_list);
+				soc_unbind_aux_dev(comp);
 			}
 		}
 	}
-- 
cgit v1.2.3-59-g8ed1b


From e8fbd2505242467044ec51bf57c642a50ed28a14 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 4 Sep 2019 09:15:40 +0900
Subject: ASoC: soc-core: self contained soc_unbind_aux_dev()

Current soc_unbind_aux_dev() implementation is very half,
thus it is very unreadable.

	for_each_comp_order(order) {
		for_each_card_auxs_safe(card, comp, _comp) {

(1)			if (comp->driver->remove_order == order) {
				...
=>				soc_unbind_aux_dev(comp);
			}
	}

soc_unbind_aux_dev() itself is not related to remove_order (1).
And, it is called from soc_remove_aux_devices(), even though
its paired function soc_bind_aux_dev() is called from
snd_soc_instantiate_card().
It is very unbalance, and very difficult to understand.

This patch do
1) update soc_bind_aux_dev() to self contained
2) call it from soc_cleanup_card_resources() to make up balance

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r24wor0z.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 17 +++++++++--------
 1 file changed, 9 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2960070e68bc..35f48e9c5ead 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1559,10 +1559,14 @@ static int soc_link_init(struct snd_soc_card *card,
 	return ret;
 }
 
-static void soc_unbind_aux_dev(struct snd_soc_component *component)
+static void soc_unbind_aux_dev(struct snd_soc_card *card)
 {
-	component->init = NULL;
-	list_del(&component->card_aux_list);
+	struct snd_soc_component *component, *_component;
+
+	for_each_card_auxs_safe(card, component, _component) {
+		component->init = NULL;
+		list_del(&component->card_aux_list);
+	}
 }
 
 static int soc_bind_aux_dev(struct snd_soc_card *card)
@@ -1614,12 +1618,8 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 
 	for_each_comp_order(order) {
 		for_each_card_auxs_safe(card, comp, _comp) {
-
-			if (comp->driver->remove_order == order) {
+			if (comp->driver->remove_order == order)
 				soc_remove_component(comp);
-				/* remove it from the card's aux_comp_list */
-				soc_unbind_aux_dev(comp);
-			}
 		}
 	}
 }
@@ -1932,6 +1932,7 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card)
 
 	/* remove auxiliary devices */
 	soc_remove_aux_devices(card);
+	soc_unbind_aux_dev(card);
 
 	snd_soc_dapm_free(&card->dapm);
 	soc_cleanup_card_debugfs(card);
-- 
cgit v1.2.3-59-g8ed1b


From 33b773dc9288eb15d3216628d1d2381103d854a9 Mon Sep 17 00:00:00 2001
From: Curtis Malainey <cujomalainey@chromium.org>
Date: Fri, 6 Sep 2019 12:46:23 -0700
Subject: ASoC: rt5677: Remove magic number register writes

In order to simplify understanding what register values are being
written to the codec for debugging more advanced features (such as
hotwording) it is best to remove magic numbers

Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20190906194636.217881-2-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 16 +++++++++++-----
 1 file changed, 11 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index c779dc3474f9..5b6ca3ced13b 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -691,10 +691,12 @@ static void rt5677_set_dsp_mode(struct snd_soc_component *component, bool on)
 	struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component);
 
 	if (on) {
-		regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x2);
+		regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1,
+			RT5677_PWR_DSP, RT5677_PWR_DSP);
 		rt5677->is_dsp_mode = true;
 	} else {
-		regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x0);
+		regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1,
+			RT5677_PWR_DSP, 0x0);
 		rt5677->is_dsp_mode = false;
 	}
 }
@@ -4466,7 +4468,8 @@ static int rt5677_set_bias_level(struct snd_soc_component *component,
 
 			regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
 				RT5677_LDO1_SEL_MASK | RT5677_LDO2_SEL_MASK,
-				0x0055);
+				5 << RT5677_LDO1_SEL_SFT |
+				5 << RT5677_LDO2_SEL_SFT);
 			regmap_update_bits(rt5677->regmap,
 				RT5677_PR_BASE + RT5677_BIAS_CUR4,
 				0x0f00, 0x0f00);
@@ -4491,7 +4494,9 @@ static int rt5677_set_bias_level(struct snd_soc_component *component,
 		regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0);
 		regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000);
 		regmap_write(rt5677->regmap, RT5677_PWR_DIG2, 0x0000);
-		regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0022);
+		regmap_write(rt5677->regmap, RT5677_PWR_ANLG1,
+			2 << RT5677_LDO1_SEL_SFT |
+			2 << RT5677_LDO2_SEL_SFT);
 		regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000);
 		regmap_update_bits(rt5677->regmap,
 			RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000);
@@ -4719,7 +4724,8 @@ static int rt5677_probe(struct snd_soc_component *component)
 
 	regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC,
 			~RT5677_IRQ_DEBOUNCE_SEL_MASK, 0x0020);
-	regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00);
+	regmap_write(rt5677->regmap, RT5677_PWR_DSP2,
+			RT5677_PWR_SLIM_ISO | RT5677_PWR_CORE_ISO);
 
 	for (i = 0; i < RT5677_GPIO_NUM; i++)
 		rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]);
-- 
cgit v1.2.3-59-g8ed1b


From dfe58f2011595e7512bde9dffbd0abfc3a736ab7 Mon Sep 17 00:00:00 2001
From: Ben Zhang <benzh@chromium.org>
Date: Fri, 6 Sep 2019 12:46:24 -0700
Subject: ASoC: rt5677: keep analog power register at SND_SOC_BIAS_OFF

Instead of clearing RT5677_PWR_ANLG2 (MX-64h) to 0 at SND_SOC_BIAS_OFF,
we only clear the RT5677_PWR_CORE bit which is set at SND_SOC_BIAS_PREPARE.
MICBIAS control bits are left unchanged.

This fixed the bug where if MICBIAS1 widget is forced on, MICBIAS
control bits will be cleared at suspend and never turned back on again,
since DAPM thinks the widget is always on.

Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20190906194636.217881-3-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5b6ca3ced13b..315a3d39bc09 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4493,11 +4493,11 @@ static int rt5677_set_bias_level(struct snd_soc_component *component,
 	case SND_SOC_BIAS_OFF:
 		regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0);
 		regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000);
-		regmap_write(rt5677->regmap, RT5677_PWR_DIG2, 0x0000);
 		regmap_write(rt5677->regmap, RT5677_PWR_ANLG1,
 			2 << RT5677_LDO1_SEL_SFT |
 			2 << RT5677_LDO2_SEL_SFT);
-		regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000);
+		regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2,
+			RT5677_PWR_CORE, 0);
 		regmap_update_bits(rt5677->regmap,
 			RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000);
 
-- 
cgit v1.2.3-59-g8ed1b


From 6fa5963c37a2e3335eba0b7455e35a01318ebc15 Mon Sep 17 00:00:00 2001
From: Cheng-Yi Chiang <cychiang@chromium.org>
Date: Wed, 17 Jul 2019 16:33:23 +0800
Subject: ASoC: hdmi-codec: Add an op to set callback function for plug event

Add an op in hdmi_codec_ops so codec driver can register callback
function to handle plug event.

Driver in DRM can use this callback function to report connector status.

Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190717083327.47646-2-cychiang@chromium.org
Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/hdmi-codec.h    | 17 ++++++++++++++++
 sound/soc/codecs/hdmi-codec.c | 46 +++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 63 insertions(+)

(limited to 'sound')

diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
index 7fea496f1f34..83b17682e01c 100644
--- a/include/sound/hdmi-codec.h
+++ b/include/sound/hdmi-codec.h
@@ -47,6 +47,9 @@ struct hdmi_codec_params {
 	int channels;
 };
 
+typedef void (*hdmi_codec_plugged_cb)(struct device *dev,
+				      bool plugged);
+
 struct hdmi_codec_pdata;
 struct hdmi_codec_ops {
 	/*
@@ -88,6 +91,14 @@ struct hdmi_codec_ops {
 	 */
 	int (*get_dai_id)(struct snd_soc_component *comment,
 			  struct device_node *endpoint);
+
+	/*
+	 * Hook callback function to handle connector plug event.
+	 * Optional
+	 */
+	int (*hook_plugged_cb)(struct device *dev, void *data,
+			       hdmi_codec_plugged_cb fn,
+			       struct device *codec_dev);
 };
 
 /* HDMI codec initalization data */
@@ -99,6 +110,12 @@ struct hdmi_codec_pdata {
 	void *data;
 };
 
+struct snd_soc_component;
+struct snd_soc_jack;
+
+int hdmi_codec_set_jack_detect(struct snd_soc_component *component,
+			       struct snd_soc_jack *jack);
+
 #define HDMI_CODEC_DRV_NAME "hdmi-audio-codec"
 
 #endif /* __HDMI_CODEC_H__ */
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 0bf1c8cad108..b5fd8f08726e 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -7,6 +7,7 @@
 #include <linux/module.h>
 #include <linux/string.h>
 #include <sound/core.h>
+#include <sound/jack.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -274,6 +275,8 @@ struct hdmi_codec_priv {
 	struct snd_pcm_chmap *chmap_info;
 	unsigned int chmap_idx;
 	struct mutex lock;
+	struct snd_soc_jack *jack;
+	unsigned int jack_status;
 };
 
 static const struct snd_soc_dapm_widget hdmi_widgets[] = {
@@ -663,6 +666,49 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
+static void hdmi_codec_jack_report(struct hdmi_codec_priv *hcp,
+				   unsigned int jack_status)
+{
+	if (hcp->jack && jack_status != hcp->jack_status) {
+		snd_soc_jack_report(hcp->jack, jack_status, SND_JACK_LINEOUT);
+		hcp->jack_status = jack_status;
+	}
+}
+
+static void plugged_cb(struct device *dev, bool plugged)
+{
+	struct hdmi_codec_priv *hcp = dev_get_drvdata(dev);
+
+	if (plugged)
+		hdmi_codec_jack_report(hcp, SND_JACK_LINEOUT);
+	else
+		hdmi_codec_jack_report(hcp, 0);
+}
+
+/**
+ * hdmi_codec_set_jack_detect - register HDMI plugged callback
+ * @component: the hdmi-codec instance
+ * @jack: ASoC jack to report (dis)connection events on
+ */
+int hdmi_codec_set_jack_detect(struct snd_soc_component *component,
+			       struct snd_soc_jack *jack)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+	int ret = -EOPNOTSUPP;
+
+	if (hcp->hcd.ops->hook_plugged_cb) {
+		hcp->jack = jack;
+		ret = hcp->hcd.ops->hook_plugged_cb(component->dev->parent,
+						    hcp->hcd.data,
+						    plugged_cb,
+						    component->dev);
+		if (ret)
+			hcp->jack = NULL;
+	}
+	return ret;
+}
+EXPORT_SYMBOL_GPL(hdmi_codec_set_jack_detect);
+
 static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai)
 {
 	struct hdmi_codec_daifmt *cf = dai->playback_dma_data;
-- 
cgit v1.2.3-59-g8ed1b


From 2617120f4de6d0423384e0e86b14c78b9de84d5a Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 10 Sep 2019 22:51:51 +0900
Subject: ALSA: firewire-tascam: handle error code when getting current source
 of clock

The return value of snd_tscm_stream_get_clock() is ignored. This commit
checks the value and handle error.

Fixes: e453df44f0d6 ("ALSA: firewire-tascam: add PCM functionality")
Cc: <stable@vger.kernel.org> # v4.4+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190910135152.29800-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/tascam/tascam-pcm.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index b5ced5415e40..2377732caa52 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -56,6 +56,9 @@ static int pcm_open(struct snd_pcm_substream *substream)
 		goto err_locked;
 
 	err = snd_tscm_stream_get_clock(tscm, &clock);
+	if (err < 0)
+		goto err_locked;
+
 	if (clock != SND_TSCM_CLOCK_INTERNAL ||
 	    amdtp_stream_pcm_running(&tscm->rx_stream) ||
 	    amdtp_stream_pcm_running(&tscm->tx_stream)) {
-- 
cgit v1.2.3-59-g8ed1b


From e1a00b5b253a4f97216b9a33199a863987075162 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 10 Sep 2019 22:51:52 +0900
Subject: ALSA: firewire-tascam: check intermediate state of clock status and
 retry

2 bytes in MSB of register for clock status is zero during intermediate
state after changing status of sampling clock in models of TASCAM FireWire
series. The duration of this state differs depending on cases. During the
state, it's better to retry reading the register for current status of
the clock.

In current implementation, the intermediate state is checked only when
getting current sampling transmission frequency, then retry reading.
This care is required for the other operations to read the register.

This commit moves the codes of check and retry into helper function
commonly used for operations to read the register.

Fixes: e453df44f0d6 ("ALSA: firewire-tascam: add PCM functionality")
Cc: <stable@vger.kernel.org> # v4.4+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190910135152.29800-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/tascam/tascam-stream.c | 42 +++++++++++++++++++++++------------
 1 file changed, 28 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
index e852e46ebe6f..ccfa92fbc145 100644
--- a/sound/firewire/tascam/tascam-stream.c
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -8,20 +8,37 @@
 #include <linux/delay.h>
 #include "tascam.h"
 
+#define CLOCK_STATUS_MASK      0xffff0000
+#define CLOCK_CONFIG_MASK      0x0000ffff
+
 #define CALLBACK_TIMEOUT 500
 
 static int get_clock(struct snd_tscm *tscm, u32 *data)
 {
+	int trial = 0;
 	__be32 reg;
 	int err;
 
-	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
-				 TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
-				 &reg, sizeof(reg), 0);
-	if (err >= 0)
+	while (trial++ < 5) {
+		err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+				TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+				&reg, sizeof(reg), 0);
+		if (err < 0)
+			return err;
+
 		*data = be32_to_cpu(reg);
+		if (*data & CLOCK_STATUS_MASK)
+			break;
 
-	return err;
+		// In intermediate state after changing clock status.
+		msleep(50);
+	}
+
+	// Still in the intermediate state.
+	if (trial >= 5)
+		return -EAGAIN;
+
+	return 0;
 }
 
 static int set_clock(struct snd_tscm *tscm, unsigned int rate,
@@ -34,7 +51,7 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate,
 	err = get_clock(tscm, &data);
 	if (err < 0)
 		return err;
-	data &= 0x0000ffff;
+	data &= CLOCK_CONFIG_MASK;
 
 	if (rate > 0) {
 		data &= 0x000000ff;
@@ -79,17 +96,14 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate,
 
 int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate)
 {
-	u32 data = 0x0;
-	unsigned int trials = 0;
+	u32 data;
 	int err;
 
-	while (data == 0x0 || trials++ < 5) {
-		err = get_clock(tscm, &data);
-		if (err < 0)
-			return err;
+	err = get_clock(tscm, &data);
+	if (err < 0)
+		return err;
 
-		data = (data & 0xff000000) >> 24;
-	}
+	data = (data & 0xff000000) >> 24;
 
 	/* Check base rate. */
 	if ((data & 0x0f) == 0x01)
-- 
cgit v1.2.3-59-g8ed1b


From d1c9e44a858d706ce0b496f559b25732e6697b0c Mon Sep 17 00:00:00 2001
From: Saiyam Doshi <saiyamdoshi.in@gmail.com>
Date: Mon, 9 Sep 2019 23:15:41 +0530
Subject: ASoC: wcd9335: remove redundant use of ret variable

All these functions declares and initializes variable ret with
'0' and without modifying 'ret' variable, it is returned.

This patch removes this redundancy and returns '0' directly.

Signed-off-by: Saiyam Doshi <saiyamdoshi.in@gmail.com>
Link: https://lore.kernel.org/r/20190909174541.GA22718@SD
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd9335.c | 24 ++++++++----------------
 1 file changed, 8 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index 03f8a94bba2f..f318403133e9 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -3022,7 +3022,6 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w,
 	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
 	struct wcd9335_codec *wcd = snd_soc_component_get_drvdata(comp);
 	struct wcd_slim_codec_dai_data *dai = &wcd->dai[w->shift];
-	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
@@ -3034,7 +3033,7 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w,
 		break;
 	}
 
-	return ret;
+	return 0;
 }
 
 static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w,
@@ -3539,7 +3538,6 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w,
 	struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev);
 	int hph_mode = wcd->hph_mode;
 	u8 dem_inp;
-	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -3579,7 +3577,7 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w,
 		break;
 	};
 
-	return ret;
+	return 0;
 }
 
 static int wcd9335_codec_lineout_dac_event(struct snd_soc_dapm_widget *w,
@@ -3607,7 +3605,6 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
 	struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev);
-	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -3621,7 +3618,7 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w,
 		break;
 	};
 
-	return ret;
+	return 0;
 }
 
 static void wcd9335_codec_hph_post_pa_config(struct wcd9335_codec *wcd,
@@ -3692,7 +3689,6 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w,
 	struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev);
 	int hph_mode = wcd->hph_mode;
 	u8 dem_inp;
-	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -3731,7 +3727,7 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w,
 		break;
 	};
 
-	return ret;
+	return 0;
 }
 
 static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w,
@@ -3741,7 +3737,6 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w,
 	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
 	struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev);
 	int hph_mode = wcd->hph_mode;
-	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -3780,7 +3775,7 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w,
 		break;
 	};
 
-	return ret;
+	return 0;
 }
 
 static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w,
@@ -3789,7 +3784,6 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
 	int vol_reg = 0, mix_vol_reg = 0;
-	int ret = 0;
 
 	if (w->reg == WCD9335_ANA_LO_1_2) {
 		if (w->shift == 7) {
@@ -3837,7 +3831,7 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w,
 		break;
 	};
 
-	return ret;
+	return 0;
 }
 
 static void wcd9335_codec_init_flyback(struct snd_soc_component *component)
@@ -3892,7 +3886,6 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w,
 	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
 	struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev);
 	int hph_mode = wcd->hph_mode;
-	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -3930,14 +3923,13 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w,
 		break;
 	};
 
-	return ret;
+	return 0;
 }
 
 static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w,
 				       struct snd_kcontrol *kc, int event)
 {
 	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
-	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
@@ -3967,7 +3959,7 @@ static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w,
 		break;
 	};
 
-	return ret;
+	return 0;
 }
 
 static irqreturn_t wcd9335_slimbus_irq(int irq, void *data)
-- 
cgit v1.2.3-59-g8ed1b


From 48118a9323ab715358bf63f5172c5c2b8cf5f455 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Wed, 11 Sep 2019 11:33:31 +0300
Subject: ASoC: dmaengine: Replace strncpy() with strscpy_pad() for pcm->name

While it is safe to use strncpy in this case, the advice is to move to
strscpy or strscpy_pad.

Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190911083331.16801-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index d93db2c2b527..5552c66ca642 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -308,9 +308,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
 			pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE;
 
 		if (rtd->pcm->streams[i].pcm->name[0] == '\0') {
-			strncpy(rtd->pcm->streams[i].pcm->name,
-				rtd->pcm->streams[i].pcm->id,
-				sizeof(rtd->pcm->streams[i].pcm->name));
+			strscpy_pad(rtd->pcm->streams[i].pcm->name,
+				    rtd->pcm->streams[i].pcm->id,
+				    sizeof(rtd->pcm->streams[i].pcm->name));
 		}
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From 63d1a3488ff58e094a7f517cf93c0250f0a3f6be Mon Sep 17 00:00:00 2001
From: Viorel Suman <viorel.suman@nxp.com>
Date: Sat, 31 Aug 2019 00:59:10 +0300
Subject: ASoC: fsl_sai: Implement set_bclk_ratio

This is to allow machine drivers to set a certain bitclk rate
which might not be exactly rate * frame size.

Cc: NXP Linux Team <linux-imx@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20190830215910.31590-1-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_sai.c | 21 +++++++++++++++++++--
 sound/soc/fsl/fsl_sai.h |  1 +
 2 files changed, 20 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 728307acab90..ef0b74693093 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -137,6 +137,16 @@ static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask,
 	return 0;
 }
 
+static int fsl_sai_set_dai_bclk_ratio(struct snd_soc_dai *dai,
+				      unsigned int ratio)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai);
+
+	sai->bclk_ratio = ratio;
+
+	return 0;
+}
+
 static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
 		int clk_id, unsigned int freq, int fsl_dir)
 {
@@ -423,8 +433,14 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		slot_width = sai->slot_width;
 
 	if (!sai->is_slave_mode) {
-		ret = fsl_sai_set_bclk(cpu_dai, tx,
-				slots * slot_width * params_rate(params));
+		if (sai->bclk_ratio)
+			ret = fsl_sai_set_bclk(cpu_dai, tx,
+					       sai->bclk_ratio *
+					       params_rate(params));
+		else
+			ret = fsl_sai_set_bclk(cpu_dai, tx,
+					       slots * slot_width *
+					       params_rate(params));
 		if (ret)
 			return ret;
 
@@ -630,6 +646,7 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
 }
 
 static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
+	.set_bclk_ratio	= fsl_sai_set_dai_bclk_ratio,
 	.set_sysclk	= fsl_sai_set_dai_sysclk,
 	.set_fmt	= fsl_sai_set_dai_fmt,
 	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index b89b0ca26053..b12cb578f6d0 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -176,6 +176,7 @@ struct fsl_sai {
 	unsigned int mclk_streams;
 	unsigned int slots;
 	unsigned int slot_width;
+	unsigned int bclk_ratio;
 
 	const struct fsl_sai_soc_data *soc_data;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
-- 
cgit v1.2.3-59-g8ed1b


From fca11622d600228bec405456f41590155b3a3eca Mon Sep 17 00:00:00 2001
From: Saiyam Doshi <saiyamdoshi.in@gmail.com>
Date: Sat, 14 Sep 2019 08:41:33 +0530
Subject: ASoC: sdm845: remove unneeded semicolon

Remove excess semicolon after closing parenthesis.

Signed-off-by: Saiyam Doshi <saiyamdoshi.in@gmail.com>
Link: https://lore.kernel.org/r/20190914031133.GA28447@SD
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/sdm845.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 882f52ed8231..28f3cef696e6 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -319,7 +319,7 @@ static void  sdm845_snd_shutdown(struct snd_pcm_substream *substream)
 			snd_soc_dai_set_sysclk(cpu_dai,
 				Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
 				0, SNDRV_PCM_STREAM_PLAYBACK);
-		};
+		}
 		break;
 
 	case SECONDARY_MI2S_TX:
-- 
cgit v1.2.3-59-g8ed1b