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-rw-r--r--MAINTAINERS12
-rw-r--r--sound/core/seq/oss/seq_oss_synth.c3
-rw-r--r--sound/pci/hda/hda_codec.c24
-rw-r--r--sound/pci/hda/hda_intel.c6
-rw-r--r--sound/pci/hda/patch_hdmi.c1
-rw-r--r--sound/pci/hda/patch_realtek.c8
-rw-r--r--sound/pci/hda/patch_via.c1
-rw-r--r--sound/soc/codecs/hdmi-codec.c2
-rw-r--r--sound/soc/fsl/imx-hdmi.c2
-rw-r--r--sound/soc/sof/intel/hda-codec.c37
-rw-r--r--sound/soc/sof/intel/hda-dsp.c9
-rw-r--r--sound/soc/sof/intel/hda.h2
-rw-r--r--sound/usb/clock.c21
-rw-r--r--sound/usb/endpoint.c9
-rw-r--r--sound/usb/implicit.c17
-rw-r--r--sound/usb/pcm.c174
-rw-r--r--sound/usb/quirks.c28
17 files changed, 196 insertions, 160 deletions
diff --git a/MAINTAINERS b/MAINTAINERS
index 705776b31c8d..f79ec98bbb29 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -14504,10 +14504,18 @@ S: Supported
F: drivers/crypto/qat/
QCOM AUDIO (ASoC) DRIVERS
-M: Patrick Lai <plai@codeaurora.org>
+M: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
M: Banajit Goswami <bgoswami@codeaurora.org>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Supported
+F: sound/soc/codecs/lpass-va-macro.c
+F: sound/soc/codecs/lpass-wsa-macro.*
+F: sound/soc/codecs/msm8916-wcd-analog.c
+F: sound/soc/codecs/msm8916-wcd-digital.c
+F: sound/soc/codecs/wcd9335.*
+F: sound/soc/codecs/wcd934x.c
+F: sound/soc/codecs/wcd-clsh-v2.*
+F: sound/soc/codecs/wsa881x.c
F: sound/soc/qcom/
QCOM IPA DRIVER
@@ -16959,7 +16967,7 @@ M: Olivier Moysan <olivier.moysan@st.com>
M: Arnaud Pouliquen <arnaud.pouliquen@st.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Maintained
-F: Documentation/devicetree/bindings/sound/st,stm32-*.txt
+F: Documentation/devicetree/bindings/iio/adc/st,stm32-*.yaml
F: sound/soc/stm/
STM32 TIMER/LPTIMER DRIVERS
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index 11554d0412f0..1b8409ec2c97 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -611,7 +611,8 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in
if (info->is_midi) {
struct midi_info minf;
- snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf);
+ if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf))
+ return -ENXIO;
inf->synth_type = SYNTH_TYPE_MIDI;
inf->synth_subtype = 0;
inf->nr_voices = 16;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 687216e74526..eec1775dfffe 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2934,7 +2934,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
snd_hdac_leave_pm(&codec->core);
}
-static int hda_codec_suspend(struct device *dev)
+static int hda_codec_runtime_suspend(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
unsigned int state;
@@ -2953,7 +2953,7 @@ static int hda_codec_suspend(struct device *dev)
return 0;
}
-static int hda_codec_resume(struct device *dev)
+static int hda_codec_runtime_resume(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
@@ -2968,16 +2968,6 @@ static int hda_codec_resume(struct device *dev)
return 0;
}
-static int hda_codec_runtime_suspend(struct device *dev)
-{
- return hda_codec_suspend(dev);
-}
-
-static int hda_codec_runtime_resume(struct device *dev)
-{
- return hda_codec_resume(dev);
-}
-
#endif /* CONFIG_PM */
#ifdef CONFIG_PM_SLEEP
@@ -2998,31 +2988,31 @@ static void hda_codec_pm_complete(struct device *dev)
static int hda_codec_pm_suspend(struct device *dev)
{
dev->power.power_state = PMSG_SUSPEND;
- return hda_codec_suspend(dev);
+ return pm_runtime_force_suspend(dev);
}
static int hda_codec_pm_resume(struct device *dev)
{
dev->power.power_state = PMSG_RESUME;
- return hda_codec_resume(dev);
+ return pm_runtime_force_resume(dev);
}
static int hda_codec_pm_freeze(struct device *dev)
{
dev->power.power_state = PMSG_FREEZE;
- return hda_codec_suspend(dev);
+ return pm_runtime_force_suspend(dev);
}
static int hda_codec_pm_thaw(struct device *dev)
{
dev->power.power_state = PMSG_THAW;
- return hda_codec_resume(dev);
+ return pm_runtime_force_resume(dev);
}
static int hda_codec_pm_restore(struct device *dev)
{
dev->power.power_state = PMSG_RESTORE;
- return hda_codec_resume(dev);
+ return pm_runtime_force_resume(dev);
}
#endif /* CONFIG_PM_SLEEP */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e4dd2ff5e473..5a50d3a46445 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2484,6 +2484,9 @@ static const struct pci_device_id azx_ids[] = {
/* CometLake-S */
{ PCI_DEVICE(0x8086, 0xa3f0),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* CometLake-R */
+ { PCI_DEVICE(0x8086, 0xf0c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Icelake */
{ PCI_DEVICE(0x8086, 0x34c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
@@ -2507,6 +2510,9 @@ static const struct pci_device_id azx_ids[] = {
/* Alderlake-S */
{ PCI_DEVICE(0x8086, 0x7ad0),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* Alderlake-P */
+ { PCI_DEVICE(0x8086, 0x51c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Elkhart Lake */
{ PCI_DEVICE(0x8086, 0x4b55),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 74d246a0dc6d..97adff0cbcab 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -4346,6 +4346,7 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi),
HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi),
HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi),
HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi),
HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index dd82ff2bd5d6..ed5b6b894dc1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6371,6 +6371,7 @@ enum {
ALC256_FIXUP_HP_HEADSET_MIC,
ALC236_FIXUP_DELL_AIO_HEADSET_MIC,
ALC282_FIXUP_ACER_DISABLE_LINEOUT,
+ ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -7808,6 +7809,12 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE
},
+ [ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_limit_int_mic_boost,
+ .chained = true,
+ .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7826,6 +7833,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
+ SND_PCI_QUIRK(0x1025, 0x1094, "Acer Aspire E5-575T", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 0ab40a8a68fb..834367dd54e1 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -113,6 +113,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
spec->codec_type = VT1708S;
spec->gen.indep_hp = 1;
spec->gen.keep_eapd_on = 1;
+ spec->gen.dac_min_mute = 1;
spec->gen.pcm_playback_hook = via_playback_pcm_hook;
spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
codec->power_save_node = 1;
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index d5fcc4db8284..0f3ac22f2cf8 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -717,7 +717,7 @@ static int hdmi_codec_set_jack(struct snd_soc_component *component,
void *data)
{
struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
- int ret = -EOPNOTSUPP;
+ int ret = -ENOTSUPP;
if (hcp->hcd.ops->hook_plugged_cb) {
hcp->jack = jack;
diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c
index ede4a9ad1054..dbbb7618351c 100644
--- a/sound/soc/fsl/imx-hdmi.c
+++ b/sound/soc/fsl/imx-hdmi.c
@@ -90,7 +90,7 @@ static int imx_hdmi_init(struct snd_soc_pcm_runtime *rtd)
}
ret = snd_soc_component_set_jack(component, &data->hdmi_jack, NULL);
- if (ret && ret != -EOPNOTSUPP) {
+ if (ret && ret != -ENOTSUPP) {
dev_err(card->dev, "Can't set HDMI Jack %d\n", ret);
return ret;
}
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 6875fa570c2c..6744318de612 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -63,16 +63,18 @@ static int hda_codec_load_module(struct hda_codec *codec)
}
/* enable controller wake up event for all codecs with jack connectors */
-void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev)
+void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable)
{
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);
+ if (enable) {
+ 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);
}
@@ -81,23 +83,18 @@ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev)
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);
+ pm_request_resume(&codec->core.dev);
}
#else
-void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {}
+void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable) {}
void hda_codec_jack_check(struct snd_sof_dev *sdev) {}
#endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */
EXPORT_SYMBOL_NS(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC);
@@ -156,7 +153,8 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
if (!hdev->bus->audio_component) {
dev_dbg(sdev->dev,
"iDisp hw present but no driver\n");
- goto error;
+ ret = -ENOENT;
+ goto out;
}
hda_priv->need_display_power = true;
}
@@ -173,24 +171,23 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
* other return codes without modification
*/
if (ret == 0)
- goto error;
+ ret = -ENOENT;
}
- return ret;
-
-error:
- snd_hdac_ext_bus_device_exit(hdev);
- return -ENOENT;
-
+out:
+ if (ret < 0) {
+ snd_hdac_device_unregister(hdev);
+ put_device(&hdev->dev);
+ }
#else
hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL);
if (!hdev)
return -ENOMEM;
ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, HDA_DEV_ASOC);
+#endif
return ret;
-#endif
}
/* Codec initialization */
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 2b001151fe37..1c5e05b88a90 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -617,7 +617,7 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
if (runtime_suspend)
- hda_codec_jack_wake_enable(sdev);
+ hda_codec_jack_wake_enable(sdev, true);
/* power down all hda link */
snd_hdac_ext_bus_link_power_down_all(bus);
@@ -683,8 +683,11 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume)
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
/* check jack status */
- if (runtime_resume)
- hda_codec_jack_check(sdev);
+ if (runtime_resume) {
+ hda_codec_jack_wake_enable(sdev, false);
+ if (sdev->system_suspend_target == SOF_SUSPEND_NONE)
+ hda_codec_jack_check(sdev);
+ }
/* turn off the links that were off before suspend */
list_for_each_entry(hlink, &bus->hlink_list, list) {
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 9ec8ae0fd649..a3b6f3e9121c 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -650,7 +650,7 @@ void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev);
*/
void hda_codec_probe_bus(struct snd_sof_dev *sdev,
bool hda_codec_use_common_hdmi);
-void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev);
+void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable);
void hda_codec_jack_check(struct snd_sof_dev *sdev);
#endif /* CONFIG_SND_SOC_SOF_HDA */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 31051f2be46d..dc68ed65e478 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -485,18 +485,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip,
const struct audioformat *fmt, int rate)
{
struct usb_device *dev = chip->dev;
- struct usb_host_interface *alts;
- unsigned int ep;
unsigned char data[3];
int err, crate;
- alts = snd_usb_get_host_interface(chip, fmt->iface, fmt->altsetting);
- if (!alts)
- return -EINVAL;
- if (get_iface_desc(alts)->bNumEndpoints < 1)
- return -EINVAL;
- ep = get_endpoint(alts, 0)->bEndpointAddress;
-
/* if endpoint doesn't have sampling rate control, bail out */
if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE))
return 0;
@@ -506,11 +497,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip,
data[2] = rate >> 16;
err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
- UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
- data, sizeof(data));
+ UAC_EP_CS_ATTR_SAMPLE_RATE << 8,
+ fmt->endpoint, data, sizeof(data));
if (err < 0) {
dev_err(&dev->dev, "%d:%d: cannot set freq %d to ep %#x\n",
- fmt->iface, fmt->altsetting, rate, ep);
+ fmt->iface, fmt->altsetting, rate, fmt->endpoint);
return err;
}
@@ -524,11 +515,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip,
err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
- UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
- data, sizeof(data));
+ UAC_EP_CS_ATTR_SAMPLE_RATE << 8,
+ fmt->endpoint, data, sizeof(data));
if (err < 0) {
dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n",
- fmt->iface, fmt->altsetting, ep);
+ fmt->iface, fmt->altsetting, fmt->endpoint);
chip->sample_rate_read_error++;
return 0; /* some devices don't support reading */
}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index fe73fe3ff2bc..8e568823c992 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -1252,6 +1252,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip,
/* If the interface has been already set up, just set EP parameters */
if (!ep->iface_ref->need_setup) {
+ /* sample rate setup of UAC1 is per endpoint, and we need
+ * to update at each EP configuration
+ */
+ if (ep->cur_audiofmt->protocol == UAC_VERSION_1) {
+ err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt,
+ ep->cur_rate);
+ if (err < 0)
+ goto unlock;
+ }
err = snd_usb_endpoint_set_params(chip, ep);
if (err < 0)
goto unlock;
diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c
index 1ac2cc6c33fb..521cc846d9d9 100644
--- a/sound/usb/implicit.c
+++ b/sound/usb/implicit.c
@@ -175,11 +175,13 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip,
ifnum, alts);
}
-/* Pioneer devices: playback and capture streams sharing the same iface/altset
+/* Playback and capture EPs on Pioneer devices share the same iface/altset,
+ * but they don't seem working with the implicit fb mode well, hence we
+ * just return as if the sync were already set up.
*/
-static int add_pioneer_implicit_fb(struct snd_usb_audio *chip,
- struct audioformat *fmt,
- struct usb_host_interface *alts)
+static int skip_pioneer_sync_ep(struct snd_usb_audio *chip,
+ struct audioformat *fmt,
+ struct usb_host_interface *alts)
{
struct usb_endpoint_descriptor *epd;
@@ -194,8 +196,7 @@ static int add_pioneer_implicit_fb(struct snd_usb_audio *chip,
(epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
USB_ENDPOINT_USAGE_IMPLICIT_FB))
return 0;
- return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1,
- alts->desc.bInterfaceNumber, alts);
+ return 1; /* don't handle with the implicit fb, just skip sync EP */
}
static int __add_generic_implicit_fb(struct snd_usb_audio *chip,
@@ -298,11 +299,11 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip,
return 1;
}
- /* Pioneer devices implicit feedback with vendor spec class */
+ /* Pioneer devices with vendor spec class */
if (attr == USB_ENDPOINT_SYNC_ASYNC &&
alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
USB_ID_VENDOR(chip->usb_id) == 0x2b73 /* Pioneer */) {
- if (add_pioneer_implicit_fb(chip, fmt, alts))
+ if (skip_pioneer_sync_ep(chip, fmt, alts))
return 1;
}
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 56079901769f..078bb4c94033 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -663,7 +663,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs,
check_fmts.bits[1] = (u32)(fp->formats >> 32);
snd_mask_intersect(&check_fmts, fmts);
if (snd_mask_empty(&check_fmts)) {
- hwc_debug(" > check: no supported format %d\n", fp->format);
+ hwc_debug(" > check: no supported format 0x%llx\n", fp->formats);
return 0;
}
/* check the channels */
@@ -775,24 +775,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params,
return apply_hw_params_minmax(it, rmin, rmax);
}
-static int hw_rule_format(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
+static int apply_hw_params_format_bits(struct snd_mask *fmt, u64 fbits)
{
- struct snd_usb_substream *subs = rule->private;
- const struct audioformat *fp;
- struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
- u64 fbits;
u32 oldbits[2];
int changed;
- hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]);
- fbits = 0;
- list_for_each_entry(fp, &subs->fmt_list, list) {
- if (!hw_check_valid_format(subs, params, fp))
- continue;
- fbits |= fp->formats;
- }
-
oldbits[0] = fmt->bits[0];
oldbits[1] = fmt->bits[1];
fmt->bits[0] &= (u32)fbits;
@@ -806,6 +793,24 @@ static int hw_rule_format(struct snd_pcm_hw_params *params,
return changed;
}
+static int hw_rule_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ const struct audioformat *fp;
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ u64 fbits;
+
+ hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]);
+ fbits = 0;
+ list_for_each_entry(fp, &subs->fmt_list, list) {
+ if (!hw_check_valid_format(subs, params, fp))
+ continue;
+ fbits |= fp->formats;
+ }
+ return apply_hw_params_format_bits(fmt, fbits);
+}
+
static int hw_rule_period_time(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
@@ -833,64 +838,92 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params,
return apply_hw_params_minmax(it, pmin, UINT_MAX);
}
-/* apply PCM hw constraints from the concurrent sync EP */
-static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime,
- struct snd_usb_substream *subs)
+/* get the EP or the sync EP for implicit fb when it's already set up */
+static const struct snd_usb_endpoint *
+get_sync_ep_from_substream(struct snd_usb_substream *subs)
{
struct snd_usb_audio *chip = subs->stream->chip;
- struct snd_usb_endpoint *ep;
const struct audioformat *fp;
- int err;
+ const struct snd_usb_endpoint *ep;
list_for_each_entry(fp, &subs->fmt_list, list) {
ep = snd_usb_get_endpoint(chip, fp->endpoint);
if (ep && ep->cur_rate)
- goto found;
+ return ep;
if (!fp->implicit_fb)
continue;
/* for the implicit fb, check the sync ep as well */
ep = snd_usb_get_endpoint(chip, fp->sync_ep);
if (ep && ep->cur_rate)
- goto found;
+ return ep;
}
- return 0;
+ return NULL;
+}
- found:
- if (!find_format(&subs->fmt_list, ep->cur_format, ep->cur_rate,
- ep->cur_channels, false, NULL)) {
- usb_audio_dbg(chip, "EP 0x%x being used, but not applicable\n",
- ep->ep_num);
+/* additional hw constraints for implicit feedback mode */
+static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ const struct snd_usb_endpoint *ep;
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+ ep = get_sync_ep_from_substream(subs);
+ if (!ep)
return 0;
- }
- usb_audio_dbg(chip, "EP 0x%x being used, using fixed params:\n",
- ep->ep_num);
- usb_audio_dbg(chip, "rate=%d, period_size=%d, periods=%d\n",
- ep->cur_rate, ep->cur_period_frames,
- ep->cur_buffer_periods);
+ hwc_debug("applying %s\n", __func__);
+ return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format));
+}
- runtime->hw.formats = subs->formats;
- runtime->hw.rate_min = runtime->hw.rate_max = ep->cur_rate;
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
- runtime->hw.periods_min = runtime->hw.periods_max =
- ep->cur_buffer_periods;
+static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ const struct snd_usb_endpoint *ep;
+ struct snd_interval *it;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- hw_rule_channels, subs,
- SNDRV_PCM_HW_PARAM_FORMAT,
- SNDRV_PCM_HW_PARAM_RATE,
- -1);
- if (err < 0)
- return err;
+ ep = get_sync_ep_from_substream(subs);
+ if (!ep)
+ return 0;
- err = snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- ep->cur_period_frames,
- ep->cur_period_frames);
- if (err < 0)
- return err;
+ hwc_debug("applying %s\n", __func__);
+ it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate);
+}
- return 1; /* notify the finding */
+static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ const struct snd_usb_endpoint *ep;
+ struct snd_interval *it;
+
+ ep = get_sync_ep_from_substream(subs);
+ if (!ep)
+ return 0;
+
+ hwc_debug("applying %s\n", __func__);
+ it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return apply_hw_params_minmax(it, ep->cur_period_frames,
+ ep->cur_period_frames);
+}
+
+static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ const struct snd_usb_endpoint *ep;
+ struct snd_interval *it;
+
+ ep = get_sync_ep_from_substream(subs);
+ if (!ep)
+ return 0;
+
+ hwc_debug("applying %s\n", __func__);
+ it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS);
+ return apply_hw_params_minmax(it, ep->cur_buffer_periods,
+ ep->cur_buffer_periods);
}
/*
@@ -899,20 +932,11 @@ static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime,
static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs)
{
- struct snd_usb_audio *chip = subs->stream->chip;
const struct audioformat *fp;
unsigned int pt, ptmin;
int param_period_time_if_needed = -1;
int err;
- mutex_lock(&chip->mutex);
- err = apply_hw_constraint_from_sync(runtime, subs);
- mutex_unlock(&chip->mutex);
- if (err < 0)
- return err;
- if (err > 0) /* found the matching? */
- goto add_extra_rules;
-
runtime->hw.formats = subs->formats;
runtime->hw.rate_min = 0x7fffffff;
@@ -957,6 +981,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
hw_rule_rate, subs,
+ SNDRV_PCM_HW_PARAM_RATE,
SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_HW_PARAM_CHANNELS,
param_period_time_if_needed,
@@ -964,9 +989,9 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
if (err < 0)
return err;
-add_extra_rules:
err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
hw_rule_channels, subs,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_HW_PARAM_RATE,
param_period_time_if_needed,
@@ -975,6 +1000,7 @@ add_extra_rules:
return err;
err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
hw_rule_format, subs,
+ SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_HW_PARAM_RATE,
SNDRV_PCM_HW_PARAM_CHANNELS,
param_period_time_if_needed,
@@ -993,6 +1019,28 @@ add_extra_rules:
return err;
}
+ /* additional hw constraints for implicit fb */
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_format_implicit_fb, subs,
+ SNDRV_PCM_HW_PARAM_FORMAT, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_rate_implicit_fb, subs,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ hw_rule_period_size_implicit_fb, subs,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS,
+ hw_rule_periods_implicit_fb, subs,
+ SNDRV_PCM_HW_PARAM_PERIODS, -1);
+ if (err < 0)
+ return err;
+
return 0;
}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 89e172642d98..e196e364cef1 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1470,30 +1470,6 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs,
subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0;
}
-
-/*
- * Pioneer DJ DJM-900NXS2
- * Device needs to know the sample rate each time substream is started
- */
-static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs)
-{
- unsigned int cur_rate = subs->data_endpoint->cur_rate;
- /* Convert sample rate value to little endian */
- u8 sr[3];
-
- sr[0] = cur_rate & 0xff;
- sr[1] = (cur_rate >> 8) & 0xff;
- sr[2] = (cur_rate >> 16) & 0xff;
-
- /* Configure device */
- usb_set_interface(subs->dev, 0, 1);
- snd_usb_ctl_msg(subs->stream->chip->dev,
- usb_rcvctrlpipe(subs->stream->chip->dev, 0),
- 0x01, 0x22, 0x0100, 0x0082, &sr, 0x0003);
-
- return 0;
-}
-
void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
const struct audioformat *fmt)
{
@@ -1504,10 +1480,6 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
set_format_emu_quirk(subs, fmt);
break;
- case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
- case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
- pioneer_djm_set_format_quirk(subs);
- break;
case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
subs->stream_offset_adj = 2;
break;
Copyright © 1996 – 2023 Jason A. Donenfeld. All Rights Reverse Engineered.