Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've had quite busy weeks in this cycle.  Looking at ALSA core, the
  significant changes are a few fixes wrt timer and sequencer ioctls
  that have been revealed by fuzzer recently.  Other than that, ASoC
  core got a few updates about DAI link handling, but these are rather
  straightforward refactoring.

  In drivers scene, ASoC received quite lots of new drivers in addition
  to bunch of updates for still ongoing Intel Skylake support and
  topology API.  HD-audio gained a new HDMI/DP hotplug notification via
  component.  FireWire got a pile of code refactoring/updates with
  SCS.1x driver integration.

  More highlights are shown below.

  [ NOTE: this contains also many commits for DRM.  This is due to the
    pull of drm stable branch into sound tree, as the base of i915 audio
    component work for HD-audio.  The highlights below don't contain
    these DRM changes, as these are supposed to be pulled via drm tree
    in anyway sooner or later.  ]

  Core:
   - Handful fixes to harden ALSA timer and sequencer ioctls against
     races reported by syzkaller fuzzer
   - Irq description string can be unique to each card; only for
     HD-audio for now

  ASoC:
   - Conversion of the array of DAI links to a list for supporting
     dynamically adding and removing DAI links
   - Topology API enhancements to make everything more component based
     and being able to specify PCM links via topology
   - Some more fixes for the topology code, though it is still not final
     and ready for enabling in production; we really need to get to the
     point where that can be done
   - A pile of changes for Intel SkyLake drivers which hopefully deliver
     some useful initial functionality for systems with this chipset,
     though there is more work still to come
   - Lots of new features and cleanups for the Renesas drivers
   - ANC support for WM5110
   - New drivers: Imagination Technologies IPs, Atmel class D speaker,
     Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
     RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
   - Rename PCM1792a driver to be generic pcm179x

  HD-Audio:
   - Use audio component for i915 HDMI/DP hotplug handling
   - On-demand binding with i915 driver
   - bdl_pos_adj parameter adjustment for Baytrail controllers
   - Enable power_save_node for CX20722; this shouldn't lead to
     regression, hopefully
   - Kabylake HDMI/DP codec support
   - Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
     machines
   - A few code refactoring

  FireWire:
   - Lots of code cleanup and refactoring
   - Integrate the support of SCS.1x devices into snd-oxfw driver;
     snd-scs1x driver is obsoleted

  USB-audio:
   - Fix possible NULL dereference at disconnection
   - A regression fix for Native Instruments devices

  Misc:
   - A few code cleanups of fm801 driver"

* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
  ALSA: timer: Code cleanup
  ALSA: timer: Harden slave timer list handling
  ALSA: hda - Add fixup for Dell Latitidue E6540
  ALSA: timer: Fix race among timer ioctls
  ALSA: hda - add codec support for Kabylake display audio codec
  ALSA: timer: Fix double unlink of active_list
  ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
  ALSA: hda - fix the headset mic detection problem for a Dell laptop
  ALSA: hda - Fix white noise on Dell Latitude E5550
  ALSA: hda_intel: add card number to irq description
  ALSA: seq: Fix race at timer setup and close
  ALSA: seq: Fix missing NULL check at remove_events ioctl
  ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
  ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
  ASoC: AMD: Add missing include file
  ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
  ALSA: usb: Add native DSD support for Oppo HA-1
  ASoC: Make aux_dev more like a generic component
  ASoC: bcm2835: cleanup includes by ordering them alphabetically
  ASoC: AMD: Manage ACP 2.x SRAM banks power
  ...

1 files changed, 319 insertions, 0 deletions

diff --git a/sound/firewire/oxfw/oxfw-spkr.c b/sound/firewire/oxfw/oxfw-spkr.c
new file mode 100644
index 000000000000..cb905af0660d
--- /dev/null
+++ b/sound/firewire/oxfw/oxfw-spkr.c
@@ -0,0 +1,319 @@
+/*
+ * oxfw-spkr.c - a part of driver for OXFW970/971 based devices
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "oxfw.h"
+
+struct fw_spkr {
+	bool mute;
+	s16 volume[6];
+	s16 volume_min;
+	s16 volume_max;
+
+	unsigned int mixer_channels;
+	u8 mute_fb_id;
+	u8 volume_fb_id;
+};
+
+enum control_action { CTL_READ, CTL_WRITE };
+enum control_attribute {
+	CTL_MIN		= 0x02,
+	CTL_MAX		= 0x03,
+	CTL_CURRENT	= 0x10,
+};
+
+static int avc_audio_feature_mute(struct fw_unit *unit, u8 fb_id, bool *value,
+				  enum control_action action)
+{
+	u8 *buf;
+	u8 response_ok;
+	int err;
+
+	buf = kmalloc(11, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
+	if (action == CTL_READ) {
+		buf[0] = 0x01;		/* AV/C, STATUS */
+		response_ok = 0x0c;	/*       STABLE */
+	} else {
+		buf[0] = 0x00;		/* AV/C, CONTROL */
+		response_ok = 0x09;	/*       ACCEPTED */
+	}
+	buf[1] = 0x08;			/* audio unit 0 */
+	buf[2] = 0xb8;			/* FUNCTION BLOCK */
+	buf[3] = 0x81;			/* function block type: feature */
+	buf[4] = fb_id;			/* function block ID */
+	buf[5] = 0x10;			/* control attribute: current */
+	buf[6] = 0x02;			/* selector length */
+	buf[7] = 0x00;			/* audio channel number */
+	buf[8] = 0x01;			/* control selector: mute */
+	buf[9] = 0x01;			/* control data length */
+	if (action == CTL_READ)
+		buf[10] = 0xff;
+	else
+		buf[10] = *value ? 0x70 : 0x60;
+
+	err = fcp_avc_transaction(unit, buf, 11, buf, 11, 0x3fe);
+	if (err < 0)
+		goto error;
+	if (err < 11) {
+		dev_err(&unit->device, "short FCP response\n");
+		err = -EIO;
+		goto error;
+	}
+	if (buf[0] != response_ok) {
+		dev_err(&unit->device, "mute command failed\n");
+		err = -EIO;
+		goto error;
+	}
+	if (action == CTL_READ)
+		*value = buf[10] == 0x70;
+
+	err = 0;
+
+error:
+	kfree(buf);
+
+	return err;
+}
+
+static int avc_audio_feature_volume(struct fw_unit *unit, u8 fb_id, s16 *value,
+				    unsigned int channel,
+				    enum control_attribute attribute,
+				    enum control_action action)
+{
+	u8 *buf;
+	u8 response_ok;
+	int err;
+
+	buf = kmalloc(12, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
+	if (action == CTL_READ) {
+		buf[0] = 0x01;		/* AV/C, STATUS */
+		response_ok = 0x0c;	/*       STABLE */
+	} else {
+		buf[0] = 0x00;		/* AV/C, CONTROL */
+		response_ok = 0x09;	/*       ACCEPTED */
+	}
+	buf[1] = 0x08;			/* audio unit 0 */
+	buf[2] = 0xb8;			/* FUNCTION BLOCK */
+	buf[3] = 0x81;			/* function block type: feature */
+	buf[4] = fb_id;			/* function block ID */
+	buf[5] = attribute;		/* control attribute */
+	buf[6] = 0x02;			/* selector length */
+	buf[7] = channel;		/* audio channel number */
+	buf[8] = 0x02;			/* control selector: volume */
+	buf[9] = 0x02;			/* control data length */
+	if (action == CTL_READ) {
+		buf[10] = 0xff;
+		buf[11] = 0xff;
+	} else {
+		buf[10] = *value >> 8;
+		buf[11] = *value;
+	}
+
+	err = fcp_avc_transaction(unit, buf, 12, buf, 12, 0x3fe);
+	if (err < 0)
+		goto error;
+	if (err < 12) {
+		dev_err(&unit->device, "short FCP response\n");
+		err = -EIO;
+		goto error;
+	}
+	if (buf[0] != response_ok) {
+		dev_err(&unit->device, "volume command failed\n");
+		err = -EIO;
+		goto error;
+	}
+	if (action == CTL_READ)
+		*value = (buf[10] << 8) | buf[11];
+
+	err = 0;
+
+error:
+	kfree(buf);
+
+	return err;
+}
+
+static int spkr_mute_get(struct snd_kcontrol *control,
+			 struct snd_ctl_elem_value *value)
+{
+	struct snd_oxfw *oxfw = control->private_data;
+	struct fw_spkr *spkr = oxfw->spec;
+
+	value->value.integer.value[0] = !spkr->mute;
+
+	return 0;
+}
+
+static int spkr_mute_put(struct snd_kcontrol *control,
+			 struct snd_ctl_elem_value *value)
+{
+	struct snd_oxfw *oxfw = control->private_data;
+	struct fw_spkr *spkr = oxfw->spec;
+	bool mute;
+	int err;
+
+	mute = !value->value.integer.value[0];
+
+	if (mute == spkr->mute)
+		return 0;
+
+	err = avc_audio_feature_mute(oxfw->unit, spkr->mute_fb_id, &mute,
+				     CTL_WRITE);
+	if (err < 0)
+		return err;
+	spkr->mute = mute;
+
+	return 1;
+}
+
+static int spkr_volume_info(struct snd_kcontrol *control,
+			    struct snd_ctl_elem_info *info)
+{
+	struct snd_oxfw *oxfw = control->private_data;
+	struct fw_spkr *spkr = oxfw->spec;
+
+	info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	info->count = spkr->mixer_channels;
+	info->value.integer.min = spkr->volume_min;
+	info->value.integer.max = spkr->volume_max;
+
+	return 0;
+}
+
+static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 };
+
+static int spkr_volume_get(struct snd_kcontrol *control,
+			   struct snd_ctl_elem_value *value)
+{
+	struct snd_oxfw *oxfw = control->private_data;
+	struct fw_spkr *spkr = oxfw->spec;
+	unsigned int i;
+
+	for (i = 0; i < spkr->mixer_channels; ++i)
+		value->value.integer.value[channel_map[i]] = spkr->volume[i];
+
+	return 0;
+}
+
+static int spkr_volume_put(struct snd_kcontrol *control,
+			   struct snd_ctl_elem_value *value)
+{
+	struct snd_oxfw *oxfw = control->private_data;
+	struct fw_spkr *spkr = oxfw->spec;
+	unsigned int i, changed_channels;
+	bool equal_values = true;
+	s16 volume;
+	int err;
+
+	for (i = 0; i < spkr->mixer_channels; ++i) {
+		if (value->value.integer.value[i] < spkr->volume_min ||
+		    value->value.integer.value[i] > spkr->volume_max)
+			return -EINVAL;
+		if (value->value.integer.value[i] !=
+		    value->value.integer.value[0])
+			equal_values = false;
+	}
+
+	changed_channels = 0;
+	for (i = 0; i < spkr->mixer_channels; ++i)
+		if (value->value.integer.value[channel_map[i]] !=
+							spkr->volume[i])
+			changed_channels |= 1 << (i + 1);
+
+	if (equal_values && changed_channels != 0)
+		changed_channels = 1 << 0;
+
+	for (i = 0; i <= spkr->mixer_channels; ++i) {
+		volume = value->value.integer.value[channel_map[i ? i - 1 : 0]];
+		if (changed_channels & (1 << i)) {
+			err = avc_audio_feature_volume(oxfw->unit,
+						  spkr->volume_fb_id, &volume,
+						  i, CTL_CURRENT, CTL_WRITE);
+			if (err < 0)
+				return err;
+		}
+		if (i > 0)
+			spkr->volume[i - 1] = volume;
+	}
+
+	return changed_channels != 0;
+}
+
+int snd_oxfw_add_spkr(struct snd_oxfw *oxfw, bool is_lacie)
+{
+	static const struct snd_kcontrol_new controls[] = {
+		{
+			.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+			.name = "PCM Playback Switch",
+			.info = snd_ctl_boolean_mono_info,
+			.get = spkr_mute_get,
+			.put = spkr_mute_put,
+		},
+		{
+			.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+			.name = "PCM Playback Volume",
+			.info = spkr_volume_info,
+			.get = spkr_volume_get,
+			.put = spkr_volume_put,
+		},
+	};
+	struct fw_spkr *spkr;
+	unsigned int i, first_ch;
+	int err;
+
+	spkr = kzalloc(sizeof(struct fw_spkr), GFP_KERNEL);
+	if (spkr == NULL)
+		return -ENOMEM;
+	oxfw->spec = spkr;
+
+	if (is_lacie) {
+		spkr->mixer_channels = 1;
+		spkr->mute_fb_id = 0x01;
+		spkr->volume_fb_id = 0x01;
+	} else {
+		spkr->mixer_channels = 6;
+		spkr->mute_fb_id = 0x01;
+		spkr->volume_fb_id = 0x02;
+	}
+
+	err = avc_audio_feature_volume(oxfw->unit, spkr->volume_fb_id,
+				       &spkr->volume_min, 0, CTL_MIN, CTL_READ);
+	if (err < 0)
+		return err;
+	err = avc_audio_feature_volume(oxfw->unit, spkr->volume_fb_id,
+				       &spkr->volume_max, 0, CTL_MAX, CTL_READ);
+	if (err < 0)
+		return err;
+
+	err = avc_audio_feature_mute(oxfw->unit, spkr->mute_fb_id, &spkr->mute,
+				     CTL_READ);
+	if (err < 0)
+		return err;
+
+	first_ch = spkr->mixer_channels == 1 ? 0 : 1;
+	for (i = 0; i < spkr->mixer_channels; ++i) {
+		err = avc_audio_feature_volume(oxfw->unit, spkr->volume_fb_id,
+					       &spkr->volume[i], first_ch + i,
+					       CTL_CURRENT, CTL_READ);
+		if (err < 0)
+			return err;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(controls); ++i) {
+		err = snd_ctl_add(oxfw->card,
+				  snd_ctl_new1(&controls[i], oxfw));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}