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[PATCH AUTOSEL 4.9 47/48] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Sasha Levin 23 Dec '20

23 Dec '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 46c3bbd9827952f92e250fa6ee30a797a4c4e17e ] The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201211131613.3271407-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ba0c3a381933e..53eb05d0722b0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -610,11 +610,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.27.0

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[PATCH AUTOSEL 4.9 22/48] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 95d02e25a3133..77c704847e4ac 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -242,21 +242,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -499,6 +484,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

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[PATCH AUTOSEL 4.14 65/66] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Sasha Levin 23 Dec '20

23 Dec '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 46c3bbd9827952f92e250fa6ee30a797a4c4e17e ] The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201211131613.3271407-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 21d9b7d96eb0f..35da926507ac3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -642,11 +642,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.27.0

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[PATCH AUTOSEL 4.14 31/66] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 6caf94581a0e8..1ed4a7c094c48 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -242,21 +242,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -499,6 +484,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

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[PATCH AUTOSEL 4.19 85/87] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Sasha Levin 23 Dec '20

23 Dec '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 46c3bbd9827952f92e250fa6ee30a797a4c4e17e ] The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201211131613.3271407-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f86b9b0a0607e..88d7b10375221 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -642,11 +642,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.27.0

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[PATCH AUTOSEL 4.19 67/87] ASoC: Intel: bytcr_rt5640: Add quirk for ARCHOS Cesium 140
by Sasha Levin 23 Dec '20

23 Dec '20

From: Chris Chiu <chiu(a)endlessos.org> [ Upstream commit 1bea2256aa96a2d7b1b576eb74e29d79edc9bea8 ] Tha ARCHOS Cesium 140 tablet has problem with the jack-sensing, thus the heaset functions are not working. Add quirk for this model to select the correct input map, jack-detect options and channel map to enable jack sensing and headset microphone. This device uses IN1 for its internal MIC and JD2 for jack-detect. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201208060414.27646-1-chiu@endlessos.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ec630127ef2f3..f23ddb9f810ae 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -431,6 +431,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.27.0

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[PATCH AUTOSEL 4.19 61/87] ALSA: rawmidi: Access runtime->avail always in spinlock
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 88a06d6fd6b369d88cec46c62db3e2604a2f50d5 ] The runtime->avail field may be accessed concurrently while some places refer to it without taking the runtime->lock spinlock, as detected by KCSAN. Usually this isn't a big problem, but for consistency and safety, we should take the spinlock at each place referencing this field. Reported-by: syzbot+a23a6f1215c84756577c(a)syzkaller.appspotmail.com Reported-by: syzbot+3d367d1df1d2b67f5c19(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20201206083527.21163-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/rawmidi.c | 49 +++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 9b26973fe697a..f4f855d7a7910 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -87,11 +87,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -960,7 +970,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1039,7 +1049,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1056,9 +1066,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1196,7 +1208,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1363,9 +1375,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1445,6 +1459,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1463,13 +1478,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1487,13 +1505,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } -- 2.27.0

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[PATCH AUTOSEL 4.19 37/87] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e4d2fcc89c306..5ff51c2983a19 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -243,21 +243,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -501,6 +486,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

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[PATCH AUTOSEL 5.4 126/130] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Sasha Levin 23 Dec '20

23 Dec '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 46c3bbd9827952f92e250fa6ee30a797a4c4e17e ] The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201211131613.3271407-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index df4771b9eff24..0faee80e28b66 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -645,11 +645,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.27.0

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[PATCH AUTOSEL 5.4 098/130] ASoC: Intel: bytcr_rt5640: Add quirk for ARCHOS Cesium 140
by Sasha Levin 23 Dec '20

23 Dec '20

From: Chris Chiu <chiu(a)endlessos.org> [ Upstream commit 1bea2256aa96a2d7b1b576eb74e29d79edc9bea8 ] Tha ARCHOS Cesium 140 tablet has problem with the jack-sensing, thus the heaset functions are not working. Add quirk for this model to select the correct input map, jack-detect options and channel map to enable jack sensing and headset microphone. This device uses IN1 for its internal MIC and JD2 for jack-detect. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201208060414.27646-1-chiu@endlessos.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6012367f6fe48..6b6749550fc4f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -422,6 +422,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.27.0

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[PATCH AUTOSEL 5.4 095/130] ASoC: Intel: cht_bsw_nau8824: Change SSP2-Codec DAI id to 0
by Sasha Levin 23 Dec '20

23 Dec '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 748e72e869718db8d735d773040bce95158c98c6 ] The snd-soc-sst-acpi driver does not care about the id specified for the SSP2-Codec DAI, but it does matter for the snd-sof-acpi driver; and when it is not 0 then the snd-sof-acpi driver does not work. Set the SSP2-Codec DAI id to 0, fixing the snd-sof-acpi driver not working on devices using the cht_bsw_nau8824 machine-driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201206122436.13553-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 501bad3976fbf..bea1372b416e2 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -218,7 +218,7 @@ static struct snd_soc_dai_link cht_dailink[] = { { /* SSP2 - Codec */ .name = "SSP2-Codec", - .id = 1, + .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, -- 2.27.0

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[PATCH AUTOSEL 5.4 087/130] ALSA: rawmidi: Access runtime->avail always in spinlock
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 88a06d6fd6b369d88cec46c62db3e2604a2f50d5 ] The runtime->avail field may be accessed concurrently while some places refer to it without taking the runtime->lock spinlock, as detected by KCSAN. Usually this isn't a big problem, but for consistency and safety, we should take the spinlock at each place referencing this field. Reported-by: syzbot+a23a6f1215c84756577c(a)syzkaller.appspotmail.com Reported-by: syzbot+3d367d1df1d2b67f5c19(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20201206083527.21163-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/rawmidi.c | 49 +++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 94db4683cfaff..6a3543b8455fc 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -72,11 +72,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -945,7 +955,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1024,7 +1034,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1041,9 +1051,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1181,7 +1193,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1370,9 +1382,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1452,6 +1466,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1470,13 +1485,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1494,13 +1512,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } -- 2.27.0

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[PATCH AUTOSEL 5.4 056/130] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 49ad4e7bb70b5..87389ab69b5ee 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -232,21 +232,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -512,6 +497,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

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[PATCH AUTOSEL 5.4 055/130] ALSA: usb-audio: Handle discrete rates properly in hw constraints
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit bc4e94aa8e72e79598e63a0b73febdcd8aeb541f ] In the current code, when the device provides the discrete sample rate tables with unusual sample rates, the driver tries to gather the whole values from the audioformat entries and create a hw-constraint rule to restrict with this single rate list. This is rather inefficient and may overlook the rates that are associated only with the certain audioformat entries. This patch improves the hw constraint setup by rewriting the existing hw_rule_rate(). The discrete sample rates (identified by rate_table and nr_rates of format entry) are checked in the existing hw_rule_rate() instead of extra rules; in the case of discrete rates, the function compares with each rate table entry and calculates the min/max values from there. For the contiguous rates, the behavior doesn't change. Along with it, snd_usb_pcm_check_knot() and snb_usb_substream rate_list field become superfluous, thus those are dropped. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 1 - sound/usb/pcm.c | 73 ++++++++++------------------------------------ sound/usb/stream.c | 1 - 3 files changed, 15 insertions(+), 60 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index d8ec5caf464de..d619e5e77a305 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -153,7 +153,6 @@ struct snd_usb_substream { u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ struct list_head fmt_list; /* format list */ - struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ spinlock_t lock; int last_frame_number; /* stored frame number */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 1a5e555002b2b..49ad4e7bb70b5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1034,27 +1034,31 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_usb_substream *subs = rule->private; struct audioformat *fp; struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - unsigned int rmin, rmax; + unsigned int rmin, rmax, r; int changed; + int i; hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max); - changed = 0; - rmin = rmax = 0; + rmin = UINT_MAX; + rmax = 0; list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; - if (changed++) { - if (rmin > fp->rate_min) - rmin = fp->rate_min; - if (rmax < fp->rate_max) - rmax = fp->rate_max; + if (fp->rate_table && fp->nr_rates) { + for (i = 0; i < fp->nr_rates; i++) { + r = fp->rate_table[i]; + if (!snd_interval_test(it, r)) + continue; + rmin = min(rmin, r); + rmax = max(rmax, r); + } } else { - rmin = fp->rate_min; - rmax = fp->rate_max; + rmin = min(rmin, fp->rate_min); + rmax = max(rmax, fp->rate_max); } } - if (!changed) { + if (rmin > rmax) { hwc_debug(" --> get empty\n"); it->empty = 1; return -EINVAL; @@ -1200,50 +1204,6 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return changed; } -/* - * If the device supports unusual bit rates, does the request meet these? - */ -static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) -{ - struct audioformat *fp; - int *rate_list; - int count = 0, needs_knot = 0; - int err; - - kfree(subs->rate_list.list); - subs->rate_list.list = NULL; - - list_for_each_entry(fp, &subs->fmt_list, list) { - if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) - return 0; - count += fp->nr_rates; - if (fp->rates & SNDRV_PCM_RATE_KNOT) - needs_knot = 1; - } - if (!needs_knot) - return 0; - - subs->rate_list.list = rate_list = - kmalloc_array(count, sizeof(int), GFP_KERNEL); - if (!subs->rate_list.list) - return -ENOMEM; - subs->rate_list.count = count; - subs->rate_list.mask = 0; - count = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - int i; - for (i = 0; i < fp->nr_rates; i++) - rate_list[count++] = fp->rate_table[i]; - } - err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &subs->rate_list); - if (err < 0) - return err; - - return 0; -} - /* * set up the runtime hardware information. @@ -1333,9 +1293,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; } - err = snd_usb_pcm_check_knot(runtime, subs); - if (err < 0) - return err; return snd_usb_autoresume(subs->stream->chip); } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d01edd5da6cf8..49c1b8a208582 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -47,7 +47,6 @@ static void free_substream(struct snd_usb_substream *subs) return; /* not initialized */ 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); } -- 2.27.0

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[PATCH AUTOSEL 5.4 051/130] ASoC: SOF: IPC: fix implicit type overflow
by Sasha Levin 23 Dec '20

23 Dec '20

From: Curtis Malainey <cujomalainey(a)chromium.org> [ Upstream commit 7c1d0e554a359cca77bfabd2a29b06f5322d172d ] Implicit values may have a length of 15bits (s16) so we need to declare the proper size so we don't get undefined behaviour. This appears to be arch and compiler dependent. This commit is to keep the headers aligned between the firmware and kernel. UBSan discovered this bug in the firmware. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201120144025.2166023-1-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/sof/header.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 10f00c08dbb7a..aae673b2bb5e2 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -30,12 +30,12 @@ /* Global Message - Generic */ #define SOF_GLB_TYPE_SHIFT 28 -#define SOF_GLB_TYPE_MASK (0xf << SOF_GLB_TYPE_SHIFT) +#define SOF_GLB_TYPE_MASK (0xfL << SOF_GLB_TYPE_SHIFT) #define SOF_GLB_TYPE(x) ((x) << SOF_GLB_TYPE_SHIFT) /* Command Message - Generic */ #define SOF_CMD_TYPE_SHIFT 16 -#define SOF_CMD_TYPE_MASK (0xfff << SOF_CMD_TYPE_SHIFT) +#define SOF_CMD_TYPE_MASK (0xfffL << SOF_CMD_TYPE_SHIFT) #define SOF_CMD_TYPE(x) ((x) << SOF_CMD_TYPE_SHIFT) /* Global Message Types */ -- 2.27.0

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[PATCH AUTOSEL 5.10 070/217] ASoC: Intel: sof_sdw: add quirk for new TigerLake-SDCA device
by Sasha Levin 23 Dec '20

23 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 488cdbd8931fe4bc7f374a8b429e81d0e4b7ac76 ] Add quirks for jack detection, rt715 DAI and number of speakers. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Link: https://lore.kernel.org/r/20201111214318.150529-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b29946eb43551..ca968901ac96f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -48,6 +48,16 @@ static int sof_sdw_quirk_cb(const struct dmi_system_id *id) } static const struct dmi_system_id sof_sdw_quirk_table[] = { + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A32") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.27.0

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[PATCH AUTOSEL 5.10 066/217] ASoC: Fix vaud18 power leakage of mt6359
by Sasha Levin 23 Dec '20

23 Dec '20

From: "Shane.Chien" <shane.chien(a)mediatek.com> [ Upstream commit 64a70744b77898a15d7a5b2b4dc0fa9523a75cde ] vaud18 is power of mt6359 audio path. It should only enable when audio is used, instead of in boot up stage. Once mt6359 audio path is enabled or disabled, vaud18 is controlled by regulator supply widget "LDO_VAUD18". Due to vaud18 is controlled by regulator dapm macro instead of regmap, the macro MT6359_LDO_VAUD18_CON0 and variable avdd_reg is no used and removed from mt6359.h. Signed-off-by: Shane.Chien <shane.chien(a)mediatek.com> Link: https://lore.kernel.org/r/1604975492-6142-2-git-send-email-shane.chien@medi… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/mt6359.c | 25 +------------------------ sound/soc/codecs/mt6359.h | 8 -------- 2 files changed, 1 insertion(+), 32 deletions(-) diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 81aafb553bdd9..64b26ab969ed8 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -1833,9 +1833,7 @@ static const struct snd_soc_dapm_widget mt6359_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("CLK_BUF", SUPPLY_SEQ_CLK_BUF, MT6359_DCXO_CW12, RG_XO_AUDIO_EN_M_SFT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY_S("LDO_VAUD18", SUPPLY_SEQ_LDO_VAUD18, - MT6359_LDO_VAUD18_CON0, - RG_LDO_VAUD18_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("LDO_VAUD18", 0, 0), SND_SOC_DAPM_SUPPLY_S("AUDGLB", SUPPLY_SEQ_AUD_GLB, MT6359_AUDDEC_ANA_CON13, RG_AUDGLB_PWRDN_VA32_SFT, 1, NULL, 0), @@ -2697,20 +2695,6 @@ static int mt6359_platform_driver_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, priv); priv->dev = &pdev->dev; - priv->avdd_reg = devm_regulator_get(&pdev->dev, "vaud18"); - if (IS_ERR(priv->avdd_reg)) { - dev_err(&pdev->dev, "%s(), have no vaud18 supply: %ld", - __func__, PTR_ERR(priv->avdd_reg)); - return PTR_ERR(priv->avdd_reg); - } - - ret = regulator_enable(priv->avdd_reg); - if (ret) { - dev_err(&pdev->dev, "%s(), failed to enable regulator!\n", - __func__); - return ret; - } - ret = mt6359_parse_dt(priv); if (ret) { dev_warn(&pdev->dev, "%s() failed to parse dts\n", __func__); @@ -2731,13 +2715,6 @@ static int mt6359_platform_driver_remove(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s(), dev name %s\n", __func__, dev_name(&pdev->dev)); - ret = regulator_disable(priv->avdd_reg); - if (ret) { - dev_err(&pdev->dev, "%s(), failed to disable regulator!\n", - __func__); - return ret; - } - return 0; } diff --git a/sound/soc/codecs/mt6359.h b/sound/soc/codecs/mt6359.h index 3792e534a91b6..e0aebf639e461 100644 --- a/sound/soc/codecs/mt6359.h +++ b/sound/soc/codecs/mt6359.h @@ -135,11 +135,6 @@ /* MT6359_DCXO_CW12 */ #define RG_XO_AUDIO_EN_M_SFT 13 -/* LDO_VAUD18_CON0 */ -#define RG_LDO_VAUD18_EN_SFT 0 -#define RG_LDO_VAUD18_EN_MASK 0x1 -#define RG_LDO_VAUD18_EN_MASK_SFT (0x1 << 0) - /* AUD_TOP_CKPDN_CON0 */ #define RG_VOW13M_CK_PDN_SFT 13 #define RG_VOW13M_CK_PDN_MASK 0x1 @@ -2132,7 +2127,6 @@ #define MT6359_DCXO_CW11 0x7a6 #define MT6359_DCXO_CW12 0x7a8 -#define MT6359_LDO_VAUD18_CON0 0x1c98 #define MT6359_GPIO_MODE0 0xcc #define MT6359_GPIO_MODE0_SET 0xce @@ -2469,7 +2463,6 @@ enum { enum { /* common */ SUPPLY_SEQ_CLK_BUF, - SUPPLY_SEQ_LDO_VAUD18, SUPPLY_SEQ_AUD_GLB, SUPPLY_SEQ_HP_PULL_DOWN, SUPPLY_SEQ_CLKSQ, @@ -2629,7 +2622,6 @@ struct mt6359_priv { int hp_gain_ctl; int hp_hifi_mode; int mtkaif_protocol; - struct regulator *avdd_reg; }; #define CODEC_MT6359_NAME "mtk-codec-mt6359" -- 2.27.0

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[PATCH AUTOSEL 5.10 013/217] ASoC: intel: sof_rt5682: Add quirk for Dooly
by Sasha Levin 23 Dec '20

23 Dec '20

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit bdd088ce5bfd32b95ab1bd90b49405e7c1f1fff5 ] This DMI product family string of this board is "Google_Hatch" so the DMI quirk will take place. However, this board is using rt1015 speaker amp instead of max98357a specified in the quirk. Therefore, we need an new DMI quirk for this board. Signed-off-by: Brent Lu <brent.lu(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201030170559.20370-3-brent.lu@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ddbb9fe7cc06b..c6c33910bc354 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -99,6 +99,24 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(1)), }, + { + /* + * Dooly is hatch family but using rt1015 amp so it + * requires a quirk before "Google_Hatch". + */ + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "Dooly"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_100FS | + SOF_RT5682_SSP_AMP(1)), + }, { .callback = sof_rt5682_quirk_cb, .matches = { -- 2.27.0

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UBSAN: shift-out-of-bounds in snd_usbmidi_get_ms_info
by syzbot 22 Dec '20

22 Dec '20

Hello, syzbot found the following issue on: HEAD commit: 8653b778 Merge tag 'clk-for-linus' of git://git.kernel.org.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=109531a3500000 kernel config: https://syzkaller.appspot.com/x/.config?x=98408202fed1f636 dashboard link: https://syzkaller.appspot.com/bug?extid=92e45ae45543f89e8c88 compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1421ec47500000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=12bfa077500000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+92e45ae45543f89e8c88(a)syzkaller.appspotmail.com usb 1-1: config 0 interface 0 altsetting 0 bulk endpoint 0x8A has invalid maxpacket 31 usb 1-1: New USB device found, idVendor=324b, idProduct=5963, bcdDevice= 0.d2 usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 usb 1-1: config 0 descriptor?? usb 1-1: MIDIStreaming interface descriptor not found ================================================================================ UBSAN: shift-out-of-bounds in sound/usb/midi.c:1928:8 shift exponent 244 is too large for 32-bit type 'int' CPU: 1 PID: 17 Comm: kworker/1:0 Not tainted 5.10.0-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x107/0x163 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:395 snd_usbmidi_get_ms_info.cold+0x83/0xd4 sound/usb/midi.c:1928 __snd_usbmidi_create+0x380/0x2320 sound/usb/midi.c:2404 snd_usb_create_quirk+0xa5/0xe0 sound/usb/quirks.c:562 usb_audio_probe+0x623/0x2ab0 sound/usb/card.c:795 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

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[PATCH 1/2] ALSA: hda/realtek: Apply jack fixup for Quanta NL3
by Chris Chiu 22 Dec '20

22 Dec '20

The Quanta NL3 laptop has both a headphone output jack and a headset jack, on the right edge of the chassis. The pin information suggests that both of these are at the Front. The PulseAudio is confused to differentiate them so one of the jack can neither get the jack sense working nor the audio output. The ALC269_FIXUP_LIFEBOOK chained with ALC269_FIXUP_QUANTA_MUTE can help to differentiate 2 jacks and get the 'Auto-Mute Mode' working correctly. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..bb1010d78717 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8013,6 +8013,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.20.1

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[PATCH] pcm: Fix a typo in SND_PCM_AUDIO_TSTAMP_TYPE_LAST definition
by Takashi Iwai 22 Dec '20

22 Dec '20

It was wrongly defined with SNDRV_ prefix. Fix it. Fixes: ea02dbd20a53 ("pcm: Add snd_pcm_audio_tstamp_type_t constants") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/pcm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/pcm.h b/include/pcm.h index cf1eea8bbc9b..e300b951c076 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -361,7 +361,7 @@ typedef enum _snd_pcm_audio_tstamp_type { SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ - SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED } snd_pcm_audio_tstamp_type_t; typedef struct _snd_pcm_audio_tstamp_config { -- 2.26.2

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[PATCH 0/2] Publish snd_pcm_audio_tstamp_type_t constants
by David Henningsson 22 Dec '20

22 Dec '20

Hi, Sorry if this message reaches you twice. Apparently I tried to send this earlier from an address which I no longer has access to. :-( So if you sent a reply I could never read it. Hopefully it'll work better this time around. Anyhow, testing of these patches are limited to running "gitcompile", which does not seem to compile the tests, is there an easy command to compile them as well? David Henningsson (2): pcm: Add snd_pcm_audio_tstamp_type_t constants test/audio_time: Make use of SND_PCM_AUDIO_TSTAMP_TYPE constants include/pcm.h | 14 ++++++++++++++ test/audio_time.c | 26 +++++++++++++------------- 2 files changed, 27 insertions(+), 13 deletions(-) -- 2.25.1

2 3

[PATCH] ALSA: usb-audio: Add implicit feeback support for the BOSS GT-1
by Mike Oliphant 22 Dec '20

22 Dec '20

The BOSS GT-1 (USB ID 0582:01d6) requires implicit feedback like other similar BOSS devices. This patch adds this support. Signed-off-by: Mike Oliphant <oliphant(a)nostatic.org> --- sound/usb/implicit.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 4e911d200..d44997e49 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -77,6 +77,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ {} /* terminator */ }; @@ -87,6 +88,7 @@ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ {} /* terminator */ }; -- 2.29.2

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[PATCH v3 0/6] Add support for Rpi4b + Cirrus Lochnagar2 and CS47L15
by Richard Fitzgerald 21 Dec '20

21 Dec '20

This set of patches provides support for using the Raspberry Pi 4b with a Cirrus Logic Lochnagar 2 audio development platform plus Cirrus Logic CS47L15 codec. Patches are needed to audio-graph-card to enable support for setting the component sysclks and plls. These are not specific to CS47L15 - several Cirrus and non-Cirrus codecs also require component clock setup, typically for larger codecs with multiple internal clocking options and clock domains. The codec sysclks and plls cannot be placed under the clock framework because they are typically I2C/SPI/Soundwire-connected peripherals and access to the registers would cause a nested get of the bus clock. The clock framework does not support this and it would result in a deadlock. Richard Fitzgerald (6): of: base: Add of_count_phandle_with_fixed_args() ASoC: audio-graph-card: Add plls and sysclks DT bindings ASoC: audio-graph-card: Support setting component plls and sysclks ASoC: madera: Allow codecs to be selected from kernel config ASoC: madera: Export clock config defines to dt-bindings ARM: dts: Add dts for RPi4b + Cirrus Logic Lochnagar2 + CS47L15 .../bindings/sound/audio-graph-card.txt | 72 ++++++ MAINTAINERS | 1 + arch/arm/boot/dts/Makefile | 1 + ...bcm2711-rpi4b-cirrus-lochnagar-cs47l15.dts | 186 ++++++++++++++ .../dts/bcm2711-rpi4b-cirrus-lochnagar.dtsi | 201 +++++++++++++++ drivers/of/base.c | 73 ++++-- include/dt-bindings/sound/madera.h | 60 +++++ include/linux/of.h | 9 + include/sound/simple_card_utils.h | 25 ++ sound/soc/codecs/Kconfig | 10 +- sound/soc/codecs/madera.h | 56 +---- sound/soc/generic/audio-graph-card.c | 16 +- sound/soc/generic/simple-card-utils.c | 236 ++++++++++++++++++ 13 files changed, 861 insertions(+), 85 deletions(-) create mode 100644 arch/arm/boot/dts/bcm2711-rpi4b-cirrus-lochnagar-cs47l15.dts create mode 100644 arch/arm/boot/dts/bcm2711-rpi4b-cirrus-lochnagar.dtsi -- 2.20.1

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[PATCH] ALSA: usb-audio: Add alias entry for ASUS PRIME TRX40 PRO-S
by Takashi Iwai 21 Dec '20

21 Dec '20

ASUS PRIME TRX40 PRO-S mobo with 0b05:1918 needs the same quirk alias for another ASUS mobo (0b05:1917) for the proper mixer mapping, etc. Add the corresponding entry. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210783 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index cb0b6582dfee..d731ca62d599 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -383,6 +383,9 @@ static const struct usb_audio_device_name usb_audio_names[] = { /* ASUS ROG Strix */ PROFILE_NAME(0x0b05, 0x1917, "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + /* ASUS PRIME TRX40 PRO-S */ + PROFILE_NAME(0x0b05, 0x1918, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), /* Dell WD15 Dock */ PROFILE_NAME(0x0bda, 0x4014, "Dell", "WD15 Dock", "Dell-WD15-Dock"), -- 2.26.2

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[PATCH 2/2] sound: Remove redundant comments
by hui yang 21 Dec '20

21 Dec '20

From: YangHui <yanghui.def(a)gmail.com> Remove redundant comments Signed-off-by: YangHui <yanghui.def(a)gmail.com> --- sound/core/init.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 764dbe6..75aec71 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -149,8 +149,6 @@ static void release_card_device(struct device *dev) * @extra_size: allocate this extra size after the main soundcard structure * @card_ret: the pointer to store the created card instance * - * Creates and initializes a soundcard structure. - * * The function allocates snd_card instance via kzalloc with the given * space for the driver to use freely. The allocated struct is stored * in the given card_ret pointer. -- 2.7.4

2 1

[PATCH] ALSA: hda/realtek: Add quirk for MSI-GP73
by Takashi Iwai 20 Dec '20

20 Dec '20

MSI-GP73 (with SSID 1462:1229) requires yet again ALC1220_FIXUP_CLEVO_P950 quirk like other MSI models. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210793 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c203eba8d9ff..5478a89d94e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2516,6 +2516,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1229, "MSI-GP73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), -- 2.26.2

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[RFC][PATCH v2 0/1] design a way to change audio Jack state by software
by Hui Wang 19 Dec '20

19 Dec '20

the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls This is the layout of folder on a machine with 2 sound cards: root@hwang4-ThinkPad-P1-Gen-3:/sys/kernel/debug# tree /sys/kernel/debug/sound-core/ /sys/kernel/debug/sound-core/ ├── card0 │ ├── HDMI!DP,pcm=10 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=11 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=3 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=7 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=8 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ └── HDMI!DP,pcm=9 Jack │ ├── jackin_inject │ └── sw_inject_enable └── card1 ├── HDMI!DP,pcm=3 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── HDMI!DP,pcm=4 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── HDMI!DP,pcm=5 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── Headphone Jack │ ├── jackin_inject │ └── sw_inject_enable ├── Headset Jack │ ├── jackin_inject │ └── sw_inject_enable └── Mic Jack ├── jackin_inject └── sw_inject_enable The sw_inject_enable has rw mode, the jackin_inject has write_only mode. If users want to inject a plugin to Headphone Jack, they could run: echo 1 > 'Headphone Jack'/sw_inject_enable /* now, this jack state can't be changed by hw events */ cat 'Headphone Jack'/sw_inject_enable ack: Headphone Jack Inject Enabled: 1 echo 1 > 'Headphone Jack'/jackin_inject /* this will inject a plugin to Headphone Jack */ After testing, run echo 0 > 'Headphone Jack'/sw_inject_enable /* the hw events will change the jack state */ Hui Wang (1): alsa: jack: implement software jack injection via debugfs include/sound/core.h | 2 + sound/core/init.c | 7 ++ sound/core/jack.c | 202 ++++++++++++++++++++++++++++++++++++------- sound/core/sound.c | 8 ++ 4 files changed, 188 insertions(+), 31 deletions(-) -- 2.25.1

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[PATCH] Fix mic sound on Jieli webcam
by Marco Giunta 19 Dec '20

19 Dec '20

Hi, recently I've bought a usb webcam with integrated mic: Jieli Technology USB PHY 2.0 (1224:2a25) The video part works well, but the mic sound is speedups, "like minions" (cit.). When I connect the camera, these are the dmesg messages: kernel: usb 1-8: current rate 0 is different from the runtime rate 8000 kernel: usb 1-8: current rate 0 is different from the runtime rate 16000 kernel: usb 1-8: current rate 0 is different from the runtime rate 44100 kernel: usb 1-8: current rate 0 is different from the runtime rate 48000 kernel: usb 1-8: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. kernel: usb 1-8: [3] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 kernel: usbcore: registered new interface driver snd-usb-audio and after a while, dmesg log is filled, every 5 seconds, with: kernel: retire_capture_urb: 84 callbacks suppressed kernel: retire_capture_urb: 1714 callbacks suppressed A guy reports on ArchLinux bug website the same problem (https://bugs.archlinux.org/task/68141?opened=12995&status%5B0%5D=) and provides a patch to fix the sound issue. I've applied the patch on kernel 5.9.13 (Fedora 33 x86_64) and now the mic works, no more minions voice effect. Now dmesg messages are only: kernel: usb 1-8: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. kernel: usb 1-8: [3] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 kernel: usbcore: registered new interface driver snd-usb-audio the retire_capture_urb messages are gone. All credits for the patch go to him but I don't know how to contact that guy nor I don't know if he has already contacted you, so my question is if you could review his patch and finally apply upstream. If you need other information or you need a tester, I'm here. Thanks, Marco --- a/sound/usb/format.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/format.c 2020-10-04 02:10:21.678685952 +0300 @@ -217,6 +217,21 @@ (chip->usb_id == USB_ID(0x041e, 0x4064) || chip->usb_id == USB_ID(0x041e, 0x4068))) rate = 8000; + + // hack for "Jieli Technology USB PHY 2.0" webcam + if (chip->usb_id == USB_ID(0x1224, 0x2a25)) { + switch (rate) { + case 8000: + fp->datainterval += 4; + break; + case 16000: + fp->datainterval += 3; + break; + default: + fp->datainterval += 1; + break; + } + } fp->rate_table[fp->nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) --- a/sound/usb/endpoint.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/endpoint.c 2020-10-04 02:09:09.471978982 +0300 @@ -882,6 +882,8 @@ if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { packs_per_ms = 8 >> ep->datainterval; max_packs_per_urb = MAX_PACKS_HS; + if (!packs_per_ms) + packs_per_ms = 1; } else { packs_per_ms = 1; max_packs_per_urb = MAX_PACKS; --- a/sound/usb/quirks.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/quirks.c 2020-10-04 02:14:04.532196519 +0300 @@ -1516,6 +1516,7 @@ case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ return true; }

2 8

[PATCH] ALSA: pcm: oss: Fix a few more UBSAN fixes
by Takashi Iwai 18 Dec '20

18 Dec '20

There are a few places that call round{up|down}_pow_of_two() with the value zero, and this causes undefined behavior warnings. Avoid calling those macros if such a nonsense value is passed; it's a minor optimization as well, as we handle it as either an error or a value to be skipped, instead. Reported-by: syzbot+33ef0b6639a8d2d42b4c(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index de1917484647..142fc751a847 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -693,6 +693,8 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_buffer_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; + if (!oss_buffer_size) + return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { if (oss_buffer_size > runtime->oss.mmap_bytes) @@ -728,17 +730,21 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - min_period_size *= oss_frame_size; - min_period_size = roundup_pow_of_two(min_period_size); - if (oss_period_size < min_period_size) - oss_period_size = min_period_size; + if (min_period_size) { + min_period_size *= oss_frame_size; + min_period_size = roundup_pow_of_two(min_period_size); + if (oss_period_size < min_period_size) + oss_period_size = min_period_size; + } max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - max_period_size *= oss_frame_size; - max_period_size = rounddown_pow_of_two(max_period_size); - if (oss_period_size > max_period_size) - oss_period_size = max_period_size; + if (max_period_size) { + max_period_size *= oss_frame_size; + max_period_size = rounddown_pow_of_two(max_period_size); + if (oss_period_size > max_period_size) + oss_period_size = max_period_size; + } oss_periods = oss_buffer_size / oss_period_size; -- 2.26.2

1 0

[PATCH] ALSA: pcm: Remove snd_pcm_lib_preallocate_dma_free()
by Lars-Peter Clausen 18 Dec '20

18 Dec '20

Since commit d4cfb30fce03 ("ALSA: pcm: Set per-card upper limit of PCM buffer allocations") snd_pcm_lib_preallocate_dma_free() is a single line function that has one caller, which is another single line function. Clean this up a bit and remove snd_pcm_lib_preallocate_dma_free() and directly call do_free_pages() from snd_pcm_lib_preallocate_free(). This is a bit less boilerplate. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/pcm_memory.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 4f03ba8ed0ae..ee6e9c5eec45 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -89,14 +89,6 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz return 0; } -/* - * release the preallocated buffer if not yet done. - */ -static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream) -{ - do_free_pages(substream->pcm->card, &substream->dma_buffer); -} - /** * snd_pcm_lib_preallocate_free - release the preallocated buffer of the specified substream. * @substream: the pcm substream instance @@ -105,7 +97,7 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream */ void snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { - snd_pcm_lib_preallocate_dma_free(substream); + do_free_pages(substream->pcm->card, &substream->dma_buffer); } /** -- 2.20.1

2 1

[PATCH 0/2] ALSA: Clear PCM buffers more properly
by Takashi Iwai 18 Dec '20

18 Dec '20

Hi, here are two small fix patches for PCM buffer allocations. Basically those address two things that have been discussed in the recent thread [1]: * Clear the remaining bytes in PCM buffer pages that are exposed via mmap * Assure the page aligned allocations for genalloc This doesn't cover the cases where non-standard PCM buffer allocation is used. Those should handle the buffer clearance by themselves. Takashi [1] https://lore.kernel.org/r/05c824e5-0c33-4182-26fa-b116a42b10d6@metafoo.de === Takashi Iwai (2): ALSA: memalloc: Align buffer allocations in page size ALSA: pcm: Clear the full allocated memory at hw_params sound/core/memalloc.c | 1 + sound/core/pcm_native.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) -- 2.26.2

2 4

[PATCH] ALSA: usb-audio: Disable sample read check if firmware doesn't give back
by Takashi Iwai 18 Dec '20

18 Dec '20

Some buggy firmware don't give the current sample rate but leaves zero. Handle this case more gracefully without warning but just skip the current rate verification from the next time. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index e940dcee792b..31051f2be46d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,12 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, } crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (!crate) { + dev_info(&dev->dev, "failed to read current rate; disabling the check\n"); + chip->sample_rate_read_error = 3; /* three strikes, see above */ + return 0; + } + if (crate != rate) { dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate); // runtime->rate = crate; -- 2.26.2

1 0

Re: [PATCH 06/18] arm64: dts: renesas: beacon: Configure Audio CODEC clocks
by Geert Uytterhoeven 18 Dec '20

18 Dec '20

Hi Adam, CC alsa-devel On Sun, Dec 13, 2020 at 7:38 PM Adam Ford <aford173(a)gmail.com> wrote: > With the newly added configurable clock options, the audio CODEC can > configure the mclk automatically. Add the reference to the versaclock. > Since the devices on I2C5 can communicate at 400KHz, let's also increase > that too > > Signed-off-by: Adam Ford <aford173(a)gmail.com> Thanks for your patch! > --- a/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > +++ b/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > @@ -424,13 +424,15 @@ &i2c0 { > > &i2c5 { > status = "okay"; > - clock-frequency = <100000>; > + clock-frequency = <400000>; > pinctrl-0 = <&i2c5_pins>; > pinctrl-names = "default"; > > codec: wm8962@1a { > compatible = "wlf,wm8962"; > reg = <0x1a>; > + clocks = <&versaclock6_bb 3>; > + clock-names = "mclk"; While the driver does get the (nameless) clock, the DT bindings lack any mention of a clocks property. It would be good to update the bindings. Note that arch/arm/boot/dts/imx6-logicpd-baseboard.dtsi and arch/arm64/boot/dts/freescale/imx8mm-beacon-baseboard.dtsi (both by your hand) use "xclk" instead of "mclk"? > DCVDD-supply = <&reg_audio>; > DBVDD-supply = <&reg_audio>; > AVDD-supply = <&reg_audio>; Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

2 4

Move SDM845 out of Qualcomm folder and add C360
by GitHub pull_request - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf pull request #70 was opened from Srinivas-Kandagatla: With driver_name fix in sdm845 machine driver https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… /proc/asound/cards get populated correctly with proper driver name and card long name. With this in place we should be able to rearrange sdm845 directory to be inline with others and reuse some of the common code. Without this all the sdm845 based machines will get a generic ucm description which will not work, one of such instance is Lenovo C360 Laptop. This patchset also add support to Lenovo C360 support Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: rsnd: don't call clk_disable_unprepare() if can't use
by Kuninori Morimoto 18 Dec '20

18 Dec '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We need to care clock accessibility, because we might can't use clock for some reasons. It sets clk_rate for each clocks when enabled. This means it doesn't have clk_rate if we can't use. We can avoid to call clk_disable_unprepare() in such case. Link: https://lore.kernel.org/r/CAMuHMdWvB+p=2JqTsO7bR8uJqKqO5A2XgXFXsVAjHk3hcxgc… Reported-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - reset adg->clk_rate[i] = 0 sound/soc/sh/rcar/adg.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index b9aacf3d3b29..abdfd9cf91e2 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -366,25 +366,27 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - int i, ret; + int i; for_each_rsnd_clk(clk, adg, i) { - ret = 0; if (enable) { - ret = clk_prepare_enable(clk); + int ret = clk_prepare_enable(clk); /* * 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]); + adg->clk_rate[i] = 0; + if (ret < 0) + dev_warn(dev, "can't use clk %d\n", i); + else + adg->clk_rate[i] = clk_get_rate(clk); } else { - clk_disable_unprepare(clk); + if (adg->clk_rate[i]) + clk_disable_unprepare(clk); + adg->clk_rate[i] = 0; } - - if (ret < 0) - dev_warn(dev, "can't use clk %d\n", i); } } -- 2.25.1

3 2

[PATCH v3] ASoC: max98373: don't access volatile registers in bias level off
by Bard Liao 18 Dec '20

18 Dec '20

We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To make userspace happy, this patch returns a cached value shich should be a valid value. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v3: - Allocate cache dynamically. v2: - Return cached values instead of 0. --- sound/soc/codecs/max98373-i2c.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373-sdw.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++--- sound/soc/codecs/max98373.h | 8 ++++++++ 4 files changed, 79 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..85f6865019d4 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -19,6 +19,12 @@ #include <sound/tlv.h> #include "max98373.h" +static const u32 max98373_i2c_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R2000_SW_RESET, 0x00}, {MAX98373_R2001_INT_RAW1, 0x00}, @@ -472,6 +478,11 @@ static struct snd_soc_dai_driver max98373_dai[] = { static int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); regcache_mark_dirty(max98373->regmap); @@ -509,6 +520,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c, { int ret = 0; int reg = 0; + int i; struct max98373_priv *max98373 = NULL; max98373 = devm_kzalloc(&i2c->dev, sizeof(*max98373), GFP_KERNEL); @@ -534,6 +546,14 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + max98373->cache_num = ARRAY_SIZE(max98373_i2c_cache_reg); + max98373->cache = devm_kcalloc(&i2c->dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_i2c_cache_reg[i]; + /* voltage/current slot & gpio configuration */ max98373_slot_config(&i2c->dev, max98373); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index ec2e79c57357..b8d471d79e93 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -23,6 +23,12 @@ struct sdw_stream_data { struct sdw_stream_runtime *sdw_stream; }; +static const u32 max98373_sdw_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R0040_SCP_INIT_STAT_1, 0x00}, {MAX98373_R0041_SCP_INIT_MASK_1, 0x00}, @@ -245,6 +251,11 @@ static const struct regmap_config max98373_sdw_regmap = { static __maybe_unused int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); @@ -757,6 +768,7 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) { struct max98373_priv *max98373; int ret; + int i; struct device *dev = &slave->dev; /* Allocate and assign private driver data structure */ @@ -768,6 +780,14 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) max98373->regmap = regmap; max98373->slave = slave; + max98373->cache_num = ARRAY_SIZE(max98373_sdw_cache_reg); + max98373->cache = devm_kcalloc(dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_sdw_cache_reg[i]; + /* Read voltage and slot configuration */ max98373_slot_config(dev, max98373); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..31d571d4fac1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); + int i; + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * Register values will be cached before suspend. The cached value + * will be a valid value and userspace will happy with that. + */ + for (i = 0; i < max98373->cache_num; i++) { + if (mc->reg == max98373->cache[i].reg) { + ucontrol->value.integer.value[0] = max98373->cache[i].val; + return 0; + } + } + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 4ab29b9d51c7..71f5a5228f34 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,11 @@ /* MAX98373_R2000_SW_RESET */ #define MAX98373_SOFT_RESET (0x1 << 0) +struct max98373_cache { + u32 reg; + u32 val; +}; + struct max98373_priv { struct regmap *regmap; int reset_gpio; @@ -212,6 +217,9 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* cache for reading a valid fake feedback value */ + struct max98373_cache *cache; + int cache_num; /* variables to support soundwire */ struct sdw_slave *slave; bool hw_init; -- 2.17.1

2 1

[PATCH v6 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 18 Dec '20

18 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Changes Since v1 and v2: -- Subject lines changed Changes Since v3: -- Patch is splitted into 2 patches Changes Since v4: -- Subject lines changed Changes Since v5: -- Removed redundant initialization of map variable in lpass-platform.c Srinivasa Rao Mandadapu (2): ASoC: qcom: Fix incorrect volatile registers ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++--------------- sound/soc/qcom/lpass-platform.c | 50 ++++++++++++++++++++++++++++------------- 2 files changed, 37 insertions(+), 33 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH] ASoC: rt711: mutex between calibration and power state changes
by shumingf＠realtek.com 18 Dec '20

18 Dec '20

From: Shuming Fan <shumingf(a)realtek.com> To avoid calibration time-out, this patch adds the mutex between calibration and power state changes Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5771c02c3459..85f744184a60 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -462,6 +462,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, unsigned int read_ll, read_rl; int i; + mutex_lock(&rt711->calibrate_mutex); + /* Can't use update bit function, so read the original value first */ addr_h = mc->reg; addr_l = mc->rreg; @@ -547,6 +549,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + mutex_unlock(&rt711->calibrate_mutex); return 0; } @@ -859,9 +863,11 @@ static int rt711_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_STANDBY: + mutex_lock(&rt711->calibrate_mutex); regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + mutex_unlock(&rt711->calibrate_mutex); break; default: -- 2.29.0

2 1

[PATCH] ASoC: Intel: haswell: Add missing pm_ops
by Cezary Rojewski 18 Dec '20

18 Dec '20

haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index c55d1239e705..c763bfeb1f38 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -189,6 +189,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.17.1

2 1

[PATCH] ASoC: meson: axg-tdm-interface: fix loopback
by Jerome Brunet 18 Dec '20

18 Dec '20

When the axg-tdm-interface was introduced, the backend DAI was marked as an endpoint when DPCM was walking the DAPM graph to find a its BE. It is no longer the case since this commit 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Because of this, when DPCM finds a BE it does everything it needs on the DAIs but it won't power up the widgets between the FE and the BE if there is no actual endpoint after the BE. On meson-axg HWs, the loopback is a special DAI of the tdm-interface BE. It is only linked to the dummy codec since there no actual HW after it. >From the DAPM perspective, the DAI has no endpoint. Because of this, the TDM decoder, which is a widget between the FE and BE is not powered up. >From the user perspective, everything seems fine but no data is produced. Connecting the Loopback DAI to a dummy DAPM endpoint solves the problem. Fixes: 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index c8664ab80d45..87cac440b369 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -467,8 +467,20 @@ static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, return ret; } +static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = { + SND_SOC_DAPM_SIGGEN("Playback Signal"), +}; + +static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = { + { "Loopback", NULL, "Playback Signal" }, +}; + static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { - .set_bias_level = axg_tdm_iface_set_bias_level, + .dapm_widgets = axg_tdm_iface_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets), + .dapm_routes = axg_tdm_iface_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes), + .set_bias_level = axg_tdm_iface_set_bias_level, }; static const struct of_device_id axg_tdm_iface_of_match[] = { -- 2.29.2

3 2

[PATCH] ASoC: meson: axg-tdmin: fix axg skew offset
by Jerome Brunet 18 Dec '20

18 Dec '20

The signal captured on from tdm decoder of the AXG SoC is incorrect. It appears amplified. The skew offset of the decoder is wrong. Setting the skew offset to 3, like the g12 and sm1 SoCs, solves and gives correct data. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdmin.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 88ed95ae886b..b4faf9d5c1aa 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -224,15 +224,6 @@ static const struct axg_tdm_formatter_ops axg_tdmin_ops = { }; static const struct axg_tdm_formatter_driver axg_tdmin_drv = { - .component_drv = &axg_tdmin_component_drv, - .regmap_cfg = &axg_tdmin_regmap_cfg, - .ops = &axg_tdmin_ops, - .quirks = &(const struct axg_tdm_formatter_hw) { - .skew_offset = 2, - }, -}; - -static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { .component_drv = &axg_tdmin_component_drv, .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, @@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = { .data = &axg_tdmin_drv, }, { .compatible = "amlogic,g12a-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, { .compatible = "amlogic,sm1-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); -- 2.29.2

2 1

[PATCH v5 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 18 Dec '20

18 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Changes Since v1 and v2: -- Subject lines changed Changes Since v3: -- Patch is splitted into 2 patches Changes Since v4: -- Subject lines changed Srinivasa Rao Mandadapu (2): ASoC: qcom: Fix incorrect volatile registers ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++---------------- sound/soc/qcom/lpass-platform.c | 46 +++++++++++++++++++++++++++++++---------- 2 files changed, 37 insertions(+), 29 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH v2] ASoC: max98373: don't access volatile registers in bias level off
by Bard Liao 18 Dec '20

18 Dec '20

We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To make userspace happy, this patch returns a cached value shich should be a valid value. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - Return cached values instead of 0. --- sound/soc/codecs/max98373-i2c.c | 8 ++++++++ sound/soc/codecs/max98373-sdw.c | 8 ++++++++ sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++--- sound/soc/codecs/max98373.h | 14 ++++++++++++++ 4 files changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..09890478b120 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -472,6 +472,11 @@ static struct snd_soc_dai_driver max98373_dai[] = { static int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); regcache_mark_dirty(max98373->regmap); @@ -534,6 +539,9 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + max98373->cache = max98373_cache; + max98373->cache_num = ARRAY_SIZE(max98373_cache); + /* voltage/current slot & gpio configuration */ max98373_slot_config(&i2c->dev, max98373); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index ec2e79c57357..0575d323df8a 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -245,6 +245,11 @@ static const struct regmap_config max98373_sdw_regmap = { static __maybe_unused int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); @@ -768,6 +773,9 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) max98373->regmap = regmap; max98373->slave = slave; + max98373->cache = max98373_cache; + max98373->cache_num = ARRAY_SIZE(max98373_cache); + /* Read voltage and slot configuration */ max98373_slot_config(dev, max98373); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..31d571d4fac1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); + int i; + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * Register values will be cached before suspend. The cached value + * will be a valid value and userspace will happy with that. + */ + for (i = 0; i < max98373->cache_num; i++) { + if (mc->reg == max98373->cache[i].reg) { + ucontrol->value.integer.value[0] = max98373->cache[i].val; + return 0; + } + } + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 4ab29b9d51c7..6206c6c3a773 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,17 @@ /* MAX98373_R2000_SW_RESET */ #define MAX98373_SOFT_RESET (0x1 << 0) +struct max98373_cache { + u32 reg; + u32 val; +}; + +struct max98373_cache max98373_cache[] = { + {MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0}, + {MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0}, + {MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0}, +}; + struct max98373_priv { struct regmap *regmap; int reset_gpio; @@ -212,6 +223,9 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* cache for reading a valid fake feedback value */ + struct max98373_cache *cache; + int cache_num; /* variables to support soundwire */ struct sdw_slave *slave; bool hw_init; -- 2.17.1

2 2

UCM fails as CardDriver may not be defined
by GitHub issues - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf issue #69 was opened from coolstar: UCM doesn't load as CardDriver isn't set on some platforms Changing https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/ucm.conf#L69 to `Directory "${CardLongName}"` resolves the issue ``` I: [pulseaudio] source.c: Created source 1 "alsa_input.platform-sound.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: [pulseaudio] source.c: alsa.resolution_bits = "16" I: [pulseaudio] source.c: device.api = "alsa" I: [pulseaudio] source.c: device.class = "sound" I: [pulseaudio] source.c: alsa.class = "generic" I: [pulseaudio] source.c: alsa.subclass = "generic-mix" I: [pulseaudio] source.c: alsa.name = "" I: [pulseaudio] source.c: alsa.id = "ADMAIF1 CIF ADMAIF1-0" I: [pulseaudio] source.c: alsa.subdevice = "0" I: [pulseaudio] source.c: alsa.subdevice_name = "subdevice #0" I: [pulseaudio] source.c: alsa.device = "0" I: [pulseaudio] source.c: alsa.card = "1" I: [pulseaudio] source.c: alsa.card_name = "tegra-snd-t210ref-mobile-rt565x" I: [pulseaudio] source.c: alsa.long_card_name = "tegra-snd-t210ref-mobile-rt565x" I: [pulseaudio] source.c: device.bus_path = "platform-sound" I: [pulseaudio] source.c: sysfs.path = "/devices/sound/sound/card1" I: [pulseaudio] source.c: device.form_factor = "internal" I: [pulseaudio] source.c: device.string = "front:1" I: [pulseaudio] source.c: device.buffering.buffer_size = "32768" I: [pulseaudio] source.c: device.buffering.fragment_size = "16384" I: [pulseaudio] source.c: device.access_mode = "mmap+timer" I: [pulseaudio] source.c: device.profile.name = "analog-stereo" I: [pulseaudio] source.c: device.profile.description = "Analog Stereo" I: [pulseaudio] source.c: device.description = "Built-in Audio Analog Stereo" I: [pulseaudio] source.c: module-udev-detect.discovered = "1" I: [pulseaudio] source.c: device.icon_name = "audio-card" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/69 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1 ] ALSA: core: memalloc: add page alignment for iram
by Robin Gong 17 Dec '20

17 Dec '20

Since mmap for userspace is based on page alignment, add page alignment for iram alloc from pool, otherwise, some good data located in the same page of dmab->area maybe touched wrongly by userspace like pulseaudio. Signed-off-by: Robin Gong <yibin.gong(a)nxp.com> --- sound/core/memalloc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 0aeeb62..0f33516 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -77,7 +77,8 @@ static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) /* Assign the pool into private_data field */ dmab->private_data = pool; - dmab->area = gen_pool_dma_alloc(pool, size, &dmab->addr); + dmab->area = gen_pool_dma_alloc_align(pool, size, &dmab->addr, + PAGE_SIZE); } /** -- 2.7.4

3 13

wallclock time for midi events without audio port
by Michael Kirchner 17 Dec '20

17 Dec '20

Hi, i try to get an absolute wallclock time for midi events using event->time.time.tv_sec. As I understand I have to add trigger time to get wall clock time. Therefor I use snd_pcm_status_get_trigger_htstamp(). If I use alsa audio via snd_pcm_open there is no problem. But: how can I get trigger time if I use only a midi port without audio? (usecase is: soft synth with alsa midi in -> jack audio out) Sincerely, Michael

1 0

utils/alsa.m4: update for newer autoconf.
by GitHub pull_request - opened 17 Dec '20

17 Dec '20

alsa-project/alsa-lib pull request #106 was opened from sezero: fixes autoconf-2.70 warnings. should be compatible down to autoconf-2.59. Request URL : https://github.com/alsa-project/alsa-lib/pull/106 Patch URL : https://github.com/alsa-project/alsa-lib/pull/106.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH alsa-lib 1/2] pcm: Add snd_pcm_audio_tstamp_type_t constants
by Takashi Iwai 17 Dec '20

17 Dec '20

From: David Henningsson <diwic(a)ubuntu.com> These are mostly a copy-paste from the kernel headers. But since functions snd_pcm_audio_tstamp_config make use of these they should be added to the public API as well. Reported-by: Alex Moon <alex.r.moon(a)gmail.com> Signed-off-by: David Henningsson <diwic(a)ubuntu.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/pcm.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/include/pcm.h b/include/pcm.h index 5b0782315585..cf1eea8bbc9b 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -350,6 +350,20 @@ typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +typedef enum _snd_pcm_audio_tstamp_type { + /** + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /**< DMA time, reported as per hw_ptr */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /**< link time reported by sample or wallclock counter, reset on startup */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +} snd_pcm_audio_tstamp_type_t; + typedef struct _snd_pcm_audio_tstamp_config { /* 5 of max 16 bits used */ unsigned int type_requested:4; -- 2.26.2

1 1

Re: [PATCH 01/01] Add VID to support native DSD reproduction on FiiO devices.
by Amadej Kastelic 17 Dec '20

17 Dec '20

Message-Id: <c759bc50134ea434bfed0a183e0ce72984f5b609.camel(a)gmail.com> Add VID to support native DSD reproduction on FiiO devices. Tested-by: Amadej Kastelic <amadejkastelic7(a)gmail.com> Signed-off-by: Emilio Moretti <emilio.moretti(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 349e1e52996d..f5fc65aef628 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x292b: /* Gustard/Ess based devices */ + case 0x2972: /* FiiO devices */ case 0x2ab6: /* T+A devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ -- 2.20.1

2 3

Dell fixed type headset platform
by Kailang 17 Dec '20

17 Dec '20

Hi Takashi, Dell has fixed type headset platform for desktop. Attach patch was add pin verb. BIOS verb table was setting for fixed type coef. Many Thanks. BR, Kailang

2 1

[PATCH v4 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 17 Dec '20

17 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Srinivasa Rao Mandadapu (2): Partially revert ASoC: qcom: Fix enabling BCLK and LRCLK in LPAIF invalid state ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++---------------- sound/soc/qcom/lpass-platform.c | 46 +++++++++++++++++++++++++++++++---------- 2 files changed, 37 insertions(+), 29 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 11

rsnd: clk-multiplier already disabled warning on Ebisu
by Geert Uytterhoeven 17 Dec '20

17 Dec '20

Hi Morimoto-san, While booting Ebisu-4D, I saw the below warnings (once). rcar_sound ec500000.sound: can't use clk 1 rsnd_adg_clk_control() failed to enable CLKB, but continues. ------------[ cut here ]------------ clk-multiplier already disabled WARNING: CPU: 1 PID: 52 at drivers/clk/clk.c:952 clk_core_disable+0x30/0x9c Modules linked in: CPU: 1 PID: 52 Comm: kworker/1:2 Not tainted 5.10.0-rc7-rcar3-initrd-00582-gadecf297acf4-dirty #183 Hardware name: Renesas Ebisu-4D board based on r8a77990 (DT) Workqueue: events deferred_probe_work_func pstate: 60000085 (nZCv daIf -PAN -UAO -TCO BTYPE=--) pc : clk_core_disable+0x30/0x9c lr : clk_core_disable+0x30/0x9c sp : ffffffc010f939f0 x29: ffffffc010f939f0 x28: 0000000000000000 x27: ffffff800949eb00 x26: ffffff80093c40f8 x25: 0000000000000001 x24: ffffffc010805fea x23: 0000000000000000 x22: ffffff80082b8810 x21: ffffff80093c4080 x20: ffffff8009368b00 x19: ffffff8009368b00 x18: 0000000000000000 x17: 000000004ef774c4 x16: 0000000000000014 x15: ffffffffffffffff x14: ffffffc010b8e8b0 x13: fffffffffff9fd47 x12: 0720072007200720 x11: fffffffffffc0000 x10: ffffffc010b8e8d8 x9 : 0720072007200720 x8 : 2079646165726c61 x7 : 0000000000000004 x6 : 00000000ffffe6ad x5 : ffffffc010f93718 x4 : 0000000000000000 x3 : 0000000000000000 x2 : 00000000ffffffff x1 : 0000000000000000 x0 : 0000000000000000 Call trace: clk_core_disable+0x30/0x9c clk_core_disable_lock+0x24/0x3c clk_disable+0x20/0x30 rsnd_adg_clk_control+0xa0/0xd4 rsnd_adg_remove+0x58/0x6c rsnd_probe+0x134/0x3cc Something else fails (-EPROBE_DEFER?), and thus rsnd_adg_remove() is called. The latter disables all clocks, including the one that failed to enable before, leading to the warning. platform_drv_probe+0x54/0xa4 really_probe+0x234/0x340 driver_probe_device+0xa0/0xb0 __device_attach_driver+0x9c/0xa8 bus_for_each_drv+0x9c/0xc4 __device_attach+0xd4/0x140 device_initial_probe+0x14/0x20 bus_probe_device+0x30/0x94 deferred_probe_work_func+0x9c/0xb0 process_one_work+0x180/0x1f0 process_scheduled_works+0x44/0x48 worker_thread+0x1ec/0x270 kthread+0xdc/0xec ret_from_fork+0x10/0x30 ---[ end trace 8d8c2a1b7537ca36 ]--- ------------[ cut here ]------------ clk-multiplier already unprepared WARNING: CPU: 1 PID: 52 at drivers/clk/clk.c:810 clk_core_unprepare+0x30/0xd0 [...] ---[ end trace 8d8c2a1b7537ca37 ]--- Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

3 5

[PATCH v2] ASoC: AMD Renoir - add DMI table to avoid the ACP mic probe (broken BIOS)
by Jaroslav Kysela 17 Dec '20

17 Dec '20

Users reported that some Lenovo AMD platforms do not have ACP microphone, but the BIOS advertises it via ACPI. This patch create a simple DMI table, where those machines with the broken BIOS can be added. The DMI description for Lenovo IdeaPad 5 and IdeaPad Flex 5 devices are added there. Also describe the dmic_acpi_check kernel module parameter in a more understandable way. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 Cc: <stable(a)kernel.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index b943e59fc302..60a3e2f442db 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -6,6 +6,7 @@ #include <linux/pci.h> #include <linux/acpi.h> +#include <linux/dmi.h> #include <linux/module.h> #include <linux/io.h> #include <linux/delay.h> @@ -20,14 +21,13 @@ module_param(acp_power_gating, int, 0644); MODULE_PARM_DESC(acp_power_gating, "Enable acp power gating"); /** - * dmic_acpi_check = -1 - Checks ACPI method to know DMIC hardware status runtime - * = 0 - Skips the DMIC device creation and returns probe failure - * = 1 - Assumes that platform has DMIC support and skips ACPI - * method check + * dmic_acpi_check = -1 - Use ACPI/DMI method to detect the DMIC hardware presence at runtime + * = 0 - Skip the DMIC device creation and return probe failure + * = 1 - Force DMIC support */ static int dmic_acpi_check = ACP_DMIC_AUTO; module_param(dmic_acpi_check, bint, 0644); -MODULE_PARM_DESC(dmic_acpi_check, "checks Dmic hardware runtime"); +MODULE_PARM_DESC(dmic_acpi_check, "Digital microphone presence (-1=auto, 0=none, 1=force)"); struct acp_dev_data { void __iomem *acp_base; @@ -163,6 +163,17 @@ static int rn_acp_deinit(void __iomem *acp_base) return 0; } +static const struct dmi_system_id rn_acp_quirk_table[] = { + { + /* Lenovo IdeaPad Flex 5 14ARE05, IdeaPad 5 15ARE05 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"), + } + }, + {} +}; + static int snd_rn_acp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -172,6 +183,7 @@ static int snd_rn_acp_probe(struct pci_dev *pci, acpi_handle handle; acpi_integer dmic_status; #endif + const struct dmi_system_id *dmi_id; unsigned int irqflags; int ret, index; u32 addr; @@ -232,6 +244,12 @@ static int snd_rn_acp_probe(struct pci_dev *pci, goto de_init; } #endif + dmi_id = dmi_first_match(rn_acp_quirk_table); + if (dmi_id && !dmi_id->driver_data) { + dev_info(&pci->dev, "ACPI settings override using DMI (ACP mic is not present)"); + ret = -ENODEV; + goto de_init; + } } adata->res = devm_kzalloc(&pci->dev, -- 2.26.2

4 3

chtrt5645: Enable Internal MIC of ECS EF20EA
by GitHub pull_request - edited 17 Dec '20

17 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was edited from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * ~~Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver.~~ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

chtrt5645: Enable Internal MIC of ECS EF20EA
by GitHub pull_request - edited 17 Dec '20

17 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was edited from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: atmel: fix spelling mistake in Kconfig "programable" -> "programmable"
by Colin King 16 Dec '20

16 Dec '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of spelling mistakes in the Kconfig help text. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/atmel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 142373ec411a..9fe9471f4514 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -143,7 +143,7 @@ config SND_MCHP_SOC_SPDIFTX - sama7g5 This S/PDIF TX driver is compliant with IEC-60958 standard and - includes programable User Data and Channel Status fields. + includes programmable User Data and Channel Status fields. config SND_MCHP_SOC_SPDIFRX tristate "Microchip ASoC driver for boards using S/PDIF RX" @@ -157,5 +157,5 @@ config SND_MCHP_SOC_SPDIFRX - sama7g5 This S/PDIF RX driver is compliant with IEC-60958 standard and - includes programable User Data and Channel Status fields. + includes programmable User Data and Channel Status fields. endif -- 2.29.2

3 2

[PATCH] ASoC: codecs: fix spelling mistake in Kconfig "comunicate" -> "communicate"
by Colin King 16 Dec '20

16 Dec '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the Kconfig help text. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ba4eb54aafcb..9bf6bfdaf11e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -457,7 +457,7 @@ config SND_SOC_ADAU7118_HW help Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter. In this mode, the device works in standalone mode which - means that there is no bus to comunicate with it. Stereo mode is not + means that there is no bus to communicate with it. Stereo mode is not supported in this mode. To compile this driver as a module, choose M here: the module -- 2.29.2

2 1

[PATCH] ASoC: SOF: Fix spelling mistake in Kconfig "ond" -> "and"
by Colin King 16 Dec '20

16 Dec '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the Kconfig help text. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 031dad5fc4c7..3e8b6c035ce3 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -122,7 +122,7 @@ config SND_SOC_SOF_DEBUG_XRUN_STOP bool "SOF stop on XRUN" help This option forces PCMs to stop on any XRUN event. This is useful to - preserve any trace data ond pipeline status prior to the XRUN. + preserve any trace data and pipeline status prior to the XRUN. Say Y if you are debugging SOF FW pipeline XRUNs. If unsure select "N". -- 2.29.2

3 2

[PATCH] ASoC: dapm: remove widget from dirty list on free
by Thomas Hebb 16 Dec '20

16 Dec '20

A widget's "dirty" list_head, much like its "list" list_head, eventually chains back to a list_head on the snd_soc_card itself. This means that the list can stick around even after the widget (or all widgets) have been freed. Currently, however, widgets that are in the dirty list when freed remain there, corrupting the entire list and leading to memory errors and undefined behavior when the list is next accessed or modified. I encountered this issue when a component failed to probe relatively late in snd_soc_bind_card(), causing it to bail out and call soc_cleanup_card_resources(), which eventually called snd_soc_dapm_free() with widgets that were still dirty from when they'd been added. Fixes: db432b414e20 ("ASoC: Do DAPM power checks only for widgets changed since last run") Cc: stable(a)vger.kernel.org Signed-off-by: Thomas Hebb <tommyhebb(a)gmail.com> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7f87b449f950..148c095df27b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2486,6 +2486,7 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) enum snd_soc_dapm_direction dir; list_del(&w->list); + list_del(&w->dirty); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both -- 2.29.2

3 2

[PATCH 0/2] ASoC: ti: Maintainer mail address change
by Peter Ujfalusi 16 Dec '20

16 Dec '20

Hi, My TI address is going to bounce after Friday (18.12.2020), switch my email address to my private one for now. Regards, Peter --- Peter Ujfalusi (2): MAINTAINERS: Update email address for TI ASoC and twl4030 codec drivers ASoC: dt-bindings: ti,j721e: Update maintainer and author information .../devicetree/bindings/sound/ti,j721e-cpb-audio.yaml | 4 +++- .../devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml | 4 +++- MAINTAINERS | 6 +++--- 3 files changed, 9 insertions(+), 5 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 5

[PATCH] ALSA: hda/realtek: Remove dummy lineout on Acer TravelMate P648/P658
by Chris Chiu 16 Dec '20

16 Dec '20

Acer TravelMate laptops P648/P658 series with codec ALC282 only have one physical jack for headset but there's a confusing lineout pin on NID 0x1b reported. Audio applications hence misunderstand that there are a speaker and a lineout, and take the lineout as the default audio output. Add a new quirk to remove the useless lineout and enable the pin 0x18 for jack sensing and headset microphone. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..e7edcdbc6ca3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6366,6 +6366,7 @@ enum { ALC287_FIXUP_HP_GPIO_LED, ALC256_FIXUP_HP_HEADSET_MIC, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, + ALC282_FIXUP_ACER_DISABLE_LINEOUT, }; static const struct hda_fixup alc269_fixups[] = { @@ -7789,6 +7790,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC282_FIXUP_ACER_DISABLE_LINEOUT] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x411111f0 }, + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8560,6 +8571,22 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60140}, {0x19, 0x04a11030}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0282, 0x1025, "Acer", ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC282_STANDARD_PINS, + {0x12, 0x90a609c0}, + {0x18, 0x03a11830}, + {0x19, 0x04a19831}, + {0x1a, 0x0481303f}, + {0x1b, 0x04211020}, + {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0282, 0x1025, "Acer", ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC282_STANDARD_PINS, + {0x12, 0x90a60940}, + {0x18, 0x03a11830}, + {0x19, 0x04a19831}, + {0x1a, 0x0481303f}, + {0x1b, 0x04211020}, + {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC282_STANDARD_PINS, {0x12, 0x90a60130}, -- 2.20.1

2 1

[PATCH] ASoC: Intel: Skylake: skl-topology: Fix OOPs ib skl_tplg_complete
by Ricardo Ribalda 15 Dec '20

15 Dec '20

If dobj->control is not initialized we end up in an OOPs during skl_tplg_complete: [ 26.553358] BUG: kernel NULL pointer dereference, address: 0000000000000078 [ 26.561151] #PF: supervisor read access in kernel mode [ 26.566897] #PF: error_code(0x0000) - not-present page [ 26.572642] PGD 0 P4D 0 [ 26.575479] Oops: 0000 [#1] PREEMPT SMP PTI [ 26.580158] CPU: 2 PID: 2082 Comm: udevd Tainted: G C 5.4.81 #4 [ 26.588232] Hardware name: HP Soraka/Soraka, BIOS Google_Soraka.10431.106.0 12/03/2019 [ 26.597082] RIP: 0010:skl_tplg_complete+0x70/0x144 [snd_soc_skl] Fixes: 2d744ecf2b98 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHL") Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> --- sound/soc/intel/skylake/skl-topology.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 40bee10b0c65..0955cbb4e918 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3619,19 +3619,20 @@ static void skl_tplg_complete(struct snd_soc_component *component) list_for_each_entry(dobj, &component->dobj_list, list) { struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct soc_enum *se; + char **texts; char chan_text[4]; - if (dobj->type != SND_SOC_DOBJ_ENUM || - dobj->control.kcontrol->put != - skl_tplg_multi_config_set_dmic) + if (dobj->type != SND_SOC_DOBJ_ENUM || !kcontrol || + kcontrol->put != skl_tplg_multi_config_set_dmic) continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { - struct snd_ctl_elem_value val; + struct snd_ctl_elem_value val = {}; if (strstr(texts[i], chan_text)) { val.value.enumerated.item[0] = i; -- 2.29.2.576.ga3fc446d84-goog

3 3

Enable ECS EF20EA's internal MIC and HDMI audio
by GitHub pull_request - edited 15 Dec '20

15 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was edited from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Enable ECS EF20EA's internel MIC and HDMI audio
by GitHub pull_request - opened 15 Dec '20

15 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was opened from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire - add support for new main capture switch and main capture volume control
by GitHub pull_request - opened 15 Dec '20

15 Dec '20

alsa-project/alsa-ucm-conf pull request #67 was opened from GoPerry: This patch was added for its new main capture switch and main capture volume control which combine the ADC 07 / ADC 27 Capture and Volume switch ,the new mute switch and volume control will be handled by codec rt715`s driver mute/volume kcontrol instead of PGA5. BugLink: https://github.com/thesofproject/linux/issues/2544 Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/67 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/67.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: Intel: fix error code cnl_set_dsp_D0()
by Dan Carpenter 14 Dec '20

14 Dec '20

Return -ETIMEDOUT if the dsp boot times out instead of returning success. Fixes: cb6a55284629 ("ASoC: Intel: cnl: Add sst library functions for cnl platform") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Not tested. sound/soc/intel/skylake/cnl-sst.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index fcd8dff27ae8..1275c149acc0 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -224,6 +224,7 @@ static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) "dsp boot timeout, status=%#x error=%#x\n", sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS), sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE)); + ret = -ETIMEDOUT; goto err; } } else { -- 2.29.2

3 2

[RFC PATCH] ASoC: pcm_dmaengine: Add support for BE DAIs
by Codrin Ciubotariu 14 Dec '20

14 Dec '20

Considering the bellow scenarios: /- Gen DMA engine -\ 48KHz ******** ******* 44.1KHz ******** ******* ********* PCM0 <-FE DAI-> * FE * <-> * DSP * <-BE DAI-> * * * * * * * Ring * * * * Ring * <-> * CPU * <-> * Codec * * buff * ******* * buff * * DAI * * DAI * 44.1KHz ******** * * * * * * PCM1 <--------------------DAI-----------------> * * * * * * ******** ******* ********* For PCM0, we have two DAI links. The first DAI link is a FE, with a DSP as a CPU DAI and a platform driver .The second DAI link is a BE DAI link, with separate CPU, codec and platform drivers. We can also notice that there are two Ring buffers: the first one used by the DSP to communicate with the user-space and the second one used to move data between FE (DSP) and BE (CPU). PCM1 is a normal DAI link, with a CPU, codec and platform driver. It is exactly the previous BE DAI link from PCM0, so the samples from user-space are copied directly into the second Ring buffer. In this scenario, the BE DAI link driver should be the same, since it is decided at runtime whether the DAI is a used as a BE or not. The generic platform driver needs to be aware of this thing. For the BE case (PCM0), some callbacks of the platform driver are not called, hence the preallocated buffer is not available. Also, the PCM runtime strcture must not be touched, since it should be only used by the FE platform driver. With these changes, the generic dmaengine can also be used as a BE platform driver. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Hello, This patch is more or less incomplete for the described scenario. This is because DMAengine's pcm->config is ignored for the BE DAI link, so runtime->hw is not updated. Also, since pcm_construct/destruct are not called, the DMA channels are allocated only if DT is used. Underrun/overrun support would also be a nice to have for the transfers involving the buffer allocated for the BE. One way to hold trach of these would be to use a substream_be->runtime different than the one used for the FE. Please share your thoughts. sound/core/pcm_dmaengine.c | 18 ++++++++++-- sound/soc/soc-generic-dmaengine-pcm.c | 40 +++++++++++++++++++++++---- 2 files changed, 50 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d059ff2b2e4..5e96bc27628d 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -137,7 +137,9 @@ static void dmaengine_pcm_dma_complete(void *arg) if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) prtd->pos = 0; - snd_pcm_period_elapsed(substream); + /* do no update period for an internal PCM */ + if (!substream->pcm->internal) + snd_pcm_period_elapsed(substream); } static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) @@ -147,6 +149,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) struct dma_async_tx_descriptor *desc; enum dma_transfer_direction direction; unsigned long flags = DMA_CTRL_ACK; + dma_addr_t addr; direction = snd_pcm_substream_to_dma_direction(substream); @@ -154,11 +157,15 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) flags |= DMA_PREP_INTERRUPT; prtd->pos = 0; + if (substream->pcm->internal) + addr = substream->dma_buffer.addr; + else + addr = substream->runtime->dma_addr; + desc = dmaengine_prep_dma_cyclic(chan, - substream->runtime->dma_addr, + addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), direction, flags); - if (!desc) return -ENOMEM; @@ -315,6 +322,11 @@ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, substream->runtime->private_data = prtd; + if (substream->pcm->internal) { + substream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_IRAM; + substream->dma_buffer.dev.dev = chan->device->dev; + } + return 0; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 9ef80a48707e..f403849cd1aa 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -97,6 +97,31 @@ static int dmaengine_pcm_hw_params(struct snd_soc_component *component, if (ret) return ret; } + if (!substream->pcm->internal) { + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + } + + /* allocate a buffer for BE DAI; for now, the buffer will have the same + * size as the buffer used by the FE + */ + if (snd_dma_alloc_pages(substream->dma_buffer.dev.type, + substream->dma_buffer.dev.dev, + params_buffer_bytes(params), + &substream->dma_buffer) < 0) + return -ENOMEM; + + return 0; +} + +static int dmaengine_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (!substream->pcm->internal) + return 0; + + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; return 0; } @@ -157,9 +182,12 @@ static int dmaengine_pcm_open(struct snd_soc_component *component, struct dma_chan *chan = pcm->chan[substream->stream]; int ret; - ret = dmaengine_pcm_set_runtime_hwparams(component, substream); - if (ret) - return ret; + /* do not touch runtime if this is an internal PCM */ + if (!substream->pcm->internal) { + ret = dmaengine_pcm_set_runtime_hwparams(component, substream); + if (ret) + return ret; + } return snd_dmaengine_pcm_open(substream, chan); } @@ -309,7 +337,7 @@ static int dmaengine_copy_user(struct snd_soc_component *component, channel * (runtime->dma_bytes / runtime->channels); int ret; - if (is_playback) + if (!substream->pcm->internal && is_playback) if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; @@ -319,7 +347,7 @@ static int dmaengine_copy_user(struct snd_soc_component *component, return ret; } - if (!is_playback) + if (!substream->pcm->internal && !is_playback) if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; @@ -332,6 +360,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component = { .open = dmaengine_pcm_open, .close = dmaengine_pcm_close, .hw_params = dmaengine_pcm_hw_params, + .hw_free = dmaengine_pcm_hw_free, .trigger = dmaengine_pcm_trigger, .pointer = dmaengine_pcm_pointer, .pcm_construct = dmaengine_pcm_new, @@ -344,6 +373,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component_process = { .close = dmaengine_pcm_close, .hw_params = dmaengine_pcm_hw_params, .trigger = dmaengine_pcm_trigger, + .hw_free = dmaengine_pcm_hw_free, .pointer = dmaengine_pcm_pointer, .copy_user = dmaengine_copy_user, .pcm_construct = dmaengine_pcm_new, -- 2.27.0

3 5

[GIT PULL] ASoC updates for v5.11
by Mark Brown 14 Dec '20

14 Dec '20

The following changes since commit aa4cb898b80a28a610e26d1513e6dd42d995c225: ASoC: rt5682: change SAR voltage threshold (2020-11-26 13:06:18 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.11 for you to fetch changes up to 460aa020f56c974a3e7e5b5378b2355fec6a2c11: Merge remote-tracking branch 'asoc/for-5.11' into asoc-next (2020-12-11 17:48:04 +0000) ---------------------------------------------------------------- ASoC: Updates for v5.11 There's a lot of changes here but mostly cleanups and driver specific things, the most user visible change is the support for boot time selection of Intel DSP firmware which will make it easier for people to move over to the preferred modern implementations in distros and other large scale deployments. This also includes a merge of the new auxillary bus which was done in anticipation of use by the Intel DSP drivers which didn't quite make it. - Lots more cleanups and simplifications from Morimoto-san. - Support for some basic DPCM systems in the audio graph card from Sameer Pujar. - Remove some old pre-DT Freescale drivers for platforms that are now DT only. - Move selection of which Intel DSP implementation to use to boot time rather than requiring it to be selected at build time. - Support for Allwinner H6 I2S, Analog Devices ADAU1372, Intel Alderlake-S, GMediatek MT8192, NXP i.MX HDMI and XCVR, Realtek RT715, Qualcomm SM8250 and simple GPIO based muxes. ---------------------------------------------------------------- Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic Alexandre Belloni (4): ASoC: adau1372: Add bindings documentation ASoC: adau1372: add missing dependencies ASoC: add simple-audio-mux binding ASoC: add simple-mux Alexandru Ardelean (2): ASoC: adau1977: remove platform data and move micbias bindings include ASoC: adau1977: convert text binding to yaml format Amadeusz Sławiński (8): ASoC: topology: Remove unused functions from topology API ASoC: topology: Remove multistep topology loading ASoC: topology: Unify all device references ASoC: topology: Change allocations to resource managed ASoC: topology: Remove empty functions ASoC: topology: Simplify remove_widget function ASoC: topology: Add missing size check ASoC: topology: Fix wrong size check Arnd Bergmann (4): ASoC: cros_ec_codec: fix uninitialized memory read ASoC: atmel: mchp-spdifrx needs COMMON_CLK ASoC: fsl_aud2htx: mark PM functions as __maybe_unused ASoC: qcom: fix QDSP6 dependencies, attempt #3 Bogdan Togorean (1): ASoc: adi: Kconfig: Remove depends on for ADI reference designs Brent Lu (3): ASoC: intel: sof_rt5682: Add support for cml_rt1015_rt5682 ASoC: intel: sof_rt5682: Add quirk for Dooly ASoC: intel: sof_rt5682: Add support for tgl_rt1011_rt5682 Cezary Rojewski (4): ASoC: pcm: DRAIN support reactivation ASoC: Intel: catpt: Optimize applying user settings ASoC: Intel: catpt: Streamline power routines across LPT and WPT ASoC: Intel: catpt: Cleanup after power routines streamlining Cheng-Yi Chiang (3): ASoC: google: dt-bindings: Add sc7180-trogdor machine bindings ASoC: qcom: sc7180: Add missing PM ops ASoC: hdmi-codec: Get ELD in before reporting plugged event Chris Chiu (1): ASoC: Intel: bytcr_rt5640: Add quirk for ARCHOS Cesium 140 Christophe Branchereau (2): ASoC: codecs/jz4770: Reset interrupt flags in bias PREPARE ASoC: codecs/jz4770: Adjust timeouts for cap-coupled outputs Chuhong Yuan (2): ASoC: jz4740-i2s: add missed checks for clk_get() ASoC: amd: change clk_get() to devm_clk_get() and add missed checks Clément Péron (6): ASoC: sun4i-i2s: Fix lrck_period computation for I2S justified mode ASoC: sun4i-i2s: Change set_chan_cfg() params ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit ASoC: sun4i-i2s: Fix sun8i volatile regs ASoC: sun4i-i2s: fix coding-style for callback definition ASoC: sun4i-i2s: Document H3 with missing RX channel possibility Codrin Ciubotariu (2): ASoC: pcm5102a: Make codec selectable ASoC: atmel-i2s: do not warn if muxclk is missing Colin Ian King (1): ASoC: qcom: fix unsigned int bitwidth compared to less than zero Curtis Malainey (1): ASoC: SOF: IPC: fix implicit type overflow Dan Carpenter (7): ASoC: qcom: sc7180: Fix some indenting in sc7180_lpass_alloc_dma_channel() ASoC: qcom: common: Fix refcounting in qcom_snd_parse_of() ASoC: qcom: sc7180: initialize the "no_headphone" variable ASoC: codecs: lpass-va-macro: remove some dead code ASoC: max98390: Fix error codes in max98390_dsm_init() ASoC: wm_adsp: remove "ctl" from list on error in wm_adsp_create_control() ASoC: mediatek: mt8183: delete some unreachable code Dave Ertman (1): Add auxiliary bus support David Lin (2): ASoC: nau8315: add codec driver ASoC: nau8315: revise the power event of EN_PIN dapm widget for symmetry Derek Fang (1): ASoC: rt1015: check the return value of regmap_read during i2c probe Dharageswari R (1): ASoC: Intel: Boards: tgl_max98373: add dpcm_capture flag for speaker_smart_amp Dmitry Baryshkov (1): ASoC: qcom: sm8250: fix HDMI audio playback Fabio Estevam (8): ASoC: wm1133-ev1: Remove unused driver ASoC: mx27vis-aic32x4: Remove unused driver ASoC: phycore-ac97: Remove unused driver ASoC: imx-mc13783: Remove unused driver ASoC: fsl: eukrea: Remove the SND_SOC_IMX_SSI selection ASoC: imx-ssi: Remove unused driver ASoC: imx-audmux: Remove unused .id_table ASoC: fsl_audmix: Remove unneeded data field Fred Oh (5): ASoC: SOF: ops: add parse_platform_ext_manifest() op ASoC: SOF: Intel: hda: define parse_platform_ext_manifest op ASoC: SOF: ext_manifest: parse cavs extra config data elem ASoC: SOF: ops: modify the signature of stall op ASoC: SOF: Intel: hda: add sof_icl_ops for ICL platforms Geert Uytterhoeven (3): ASoC: fsl: SND_SOC_FSL_AUD2HTX should depend on ARCH_MXC ASoC: intel: SND_SOC_INTEL_KEEMBAY should depend on ARCH_KEEMBAY ASoC: Fix 7/8 spaces indentation in Kconfig Greg Kroah-Hartman (3): driver core: auxiliary bus: move slab.h from include file driver core: auxiliary bus: make remove function return void driver core: auxiliary bus: minor coding style tweaks Gustavo A. R. Silva (1): ASoC: codecs: Fix fall-through warnings for Clang Hans de Goede (2): ASoC: Intel: cht_bsw_nau8824: Drop compress-cpu-dai bits ASoC: Intel: cht_bsw_nau8824: Change SSP2-Codec DAI id to 0 Jack Yu (3): ASoC: rt1015: modification for calibration to get better performance ASoC/SoundWire: rt715-sdca: First version of rt715 sdw sdca codec driver ASoC: rt715: remove unused parameter Jaroslav Kysela (2): ASoC: AMD Raven/Renoir - fix the PCI probe (PCI revision) ASoC: AMD Renoir - add DMI table to avoid the ACP mic probe (broken BIOS) Jaska Uimonen (1): ASoC: SOF: control: override volume info callback Jernej Skrabec (2): ASoC: sun4i-i2s: Add support for H6 I2S ASoC: sun4i-i2s: Add H6 compatible Jerome Brunet (1): ASoC: meson: fix COMPILE_TEST error Jiaxin Yu (13): ASoC: mediatek: mt6359: add the calibration functions ASoC: mediatek: mt8192: add platform driver ASoC: mediatek: mt8192: support i2s in platform driver ASoC: mediatek: mt8192: support adda in platform driver ASoC: mediatek: mt8192: support pcm in platform driver ASoC: mediatek: mt8192: support tdm in platform driver dt-bindings: mediatek: mt8192: add audio afe document ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682 dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682 document ASoC: mediatek: mt8192: skip first time data at the beginning of DMIC recording ASoC: mt8192: revert "add audio afe document" ASoC: mediatek: mt6359: Fix regulator_dev_lookup() fails for id "LDO_VAUD18" ASoC: mt6359: remove unused property for mt6359 Jon Hunter (1): ASoC: tegra: Don't warn on probe deferral Julia Lawall (6): ASoC: wm8350: use semicolons rather than commas to separate statements ASoC: Intel: bytcr_rt5651: use semicolons rather than commas to separate statements ASoC: SOF: Intel: hda: use semicolons rather than commas to separate statements ASoC: samsung: snow: use semicolons rather than commas to separate statements ASoC: madera: use semicolons rather than commas to separate statements ASoC: dapm: use semicolons rather than commas to separate statements Kai Vehmanen (6): ASoC: SOF: loader: do not warn about unknown firmware headers ASoC: SOF: Intel: initial support for Alderlake-S ASoC: SOF: relax PCM period and buffer size constraints ASoC: SOF: Intel: add hw specific PCM constraints ASoC: Intel: common: add ACPI matching tables for Alder Lake ASoC: SOF: Intel: add SoundWire support for ADL-S Kaixu Xia (1): ASoC: wcd9335: Remove unnecessary conversion to bool Karol Trzcinski (5): ASoC: SOF: ext_manifest: Parse firmware config dictionary ASoC: SOF: Improve code alignment in header.h ASoC: SOF: Change section comment for SOF_IPC_TEST_ ASoC: SOF: Add `memory_info` file to debugfs ASoC: SOF: trace: Add runtime trace filtering mechanism Kirill Marinushkin (4): ASoC: pcm512x: Fix not setting word length if DAIFMT_CBS_CFS ASoC: pcm512x: Rearrange operations in `hw_params()` ASoC: pcm512x: Move format check into `set_fmt()` ASoC: pcm512x: Add support for more data formats Krzysztof Kozlowski (43): ASoC: sunxi: do not select COMMON_CLK to fix builds ASoC: sh: depend on COMMON_CLK to fix compile tests ASoC: stm: depend on COMMON_CLK to fix compile tests ASoC: ti: depend on COMMON_CLK to fix compile tests ASoC: ak5558: mark OF related data as maybe unused ASoC: bd28623: mark OF related data as maybe unused ASoC: gtm601: mark OF related data as maybe unused ASoC: inno_rk3036: mark OF related data as maybe unused ASoC: rk3328: mark OF related data as maybe unused ASoC: tas571x: mark OF related data as maybe unused ASoC: kirkwood: armada-370-db: mark OF related data as maybe unused ASoC: meson: t9015: mark OF related data as maybe unused ASoC: qcom: mark OF related data as maybe unused ASoC: samsung: smdk_wm8994: mark OF related data as maybe unused ASoC: rockchip: mark OF related data as maybe unused ASoC: ti: davinci: mark OF related data as maybe unused ASoC: uniphier: mark OF related data as maybe unused ASoC: ak4118: skip of_device_id table when !CONFIG_OF ASoC: alc5623: skip of_device_id table when !CONFIG_OF ASoC: alc5632: skip of_device_id table when !CONFIG_OF ASoC: da7218: skip of_device_id table when !CONFIG_OF ASoC: da7219: skip of_device_id table when !CONFIG_OF ASoC: da9055: skip of_device_id table when !CONFIG_OF ASoC: es8316: skip of_device_id table when !CONFIG_OF ASoC: max98090: skip of_device_id table when !CONFIG_OF ASoC: max98095: skip of_device_id table when !CONFIG_OF ASoC: max98371: skip of_device_id table when !CONFIG_OF ASoC: max9867: skip of_device_id table when !CONFIG_OF ASoC: max98925: skip of_device_id table when !CONFIG_OF ASoC: max98926: skip of_device_id table when !CONFIG_OF ASoC: pcm1789: skip of_device_id table when !CONFIG_OF ASoC: pcm179x: skip of_device_id table when !CONFIG_OF ASoC: rt5660: skip of_device_id table when !CONFIG_OF ASoC: tas2562: skip of_device_id table when !CONFIG_OF ASoC: tlv320: skip of_device_id table when !CONFIG_OF ASoC: ts3a227e: skip of_device_id table when !CONFIG_OF ASoC: es7134: mark OF related data as maybe unused ASoC: es7241: mark OF related data as maybe unused ASoC: samsung: i2s: mark OF related data as maybe unused ASoC: max98371: drop driver pm=NULL assignment ASoC: max98925: drop driver pm=NULL assignment ASoC: max98926: drop driver pm=NULL assignment ASoC: samsung: smdk_wm8994: remove redundant of_match_ptr() Kuninori Morimoto (34): ASoC: soc.h: remove for_each_rtd_dais_rollback() ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params() ASoC: soc-link: add mark for snd_soc_link_hw_params/free() ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free() ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free() ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free() ASoC: audio-graph-card: switch to yaml base Documentation ASoC: renesas, rsnd: switch to yaml base Documentation ASoC: renesas,rsnd: Add r8a77961 support ASoC: soc-compress: tidyup STREAM vs COMPRESS ASoC: soc-compress: assume SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx are same ASoC: soc-component: add snd_soc_component_compr_open() ASoC: soc-component: add snd_soc_component_compr_free() ASoC: soc-component: add snd_soc_component_compr_trigger() ASoC: soc-component: add snd_soc_component_compr_set_params() ASoC: soc-component: add snd_soc_component_compr_get_params() ASoC: soc-component: add snd_soc_component_compr_get_caps() ASoC: soc-component: add snd_soc_component_compr_get_codec_caps() ASoC: soc-component: add snd_soc_component_compr_ack() ASoC: soc-component: add snd_soc_component_compr_pointer() ASoC: soc-component: add snd_soc_component_compr_copy() ASoC: soc-component: add snd_soc_component_compr_set_metadata() ASoC: soc-component: add snd_soc_component_compr_get_metadata() ASoC: soc-compress: move soc_compr_free() next to soc_compr_open() ASoC: soc-dai: add mark for snd_soc_dai_compr_startup/shutdown() ASoC: soc-component: add mark for snd_soc_component_compr_open/free() ASoC: soc-component: add mark for snd_soc_link_compr_startup/shutdown() ASoC: soc-compress: add soc_compr_clean() and call it from soc_compr_open/free() ASoC: soc-core: add soc_playback_digital_mute() ASoC: soc-core: add soc_dapm_suspend_resume() ASoC: soc-core: tidyup jack.h ASoC: add soc-jack.h ASoC: soc-pcm: remove dpcm_do_trigger() ASoC: soc-pcm: care trigger rollback Kyle Russell (2): ASoC: mmp-sspa: clear transmit phase bit for non-stereo formats ASoC: mmp-sspa: set phase two word length register Lars-Peter Clausen (2): ASoC: jz4740-i2s: Remove manual DMA peripheral ID assignment ASoC: Add ADAU1372 audio CODEC support Lumi Lee (1): ASoC: mediatek: btcvsd fix tx stream assign Marcus Cooper (2): ASoC: sun4i-i2s: Set sign extend sample ASoC: sun4i-i2s: Add 20 and 24 bit support Mark Brown (48): Merge existing fixes from asoc/for-5.10 Merge series "dt-bindings: stm32: convert audio dfsdm to json-schema" from Olivier Moysan <olivier.moysan(a)st.com>: Merge series "ASoC: sun8i-codec: support for AIF2 and AIF3" from Samuel Holland <samuel(a)sholland.org>: Merge series "use semicolons rather than commas to separate statements" from Julia Lawall <Julia.Lawall(a)inria.fr>: Merge series "ASoC: qcom: add support for QRB5165 RB5 machine" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: merge soc_pcm_hw_param() rollback and soc_pcm_hw_free()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "DAI driver for new XCVR IP" from "Viorel Suman (OSS)" <viorel.suman(a)oss.nxp.com> Viorel Suman <viorel.suman(a)nxp.com>: Merge series "ASoC: SOF: cleanups for 5.10" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "Add documentation and machine driver for SC7180 sound card" from Cheng-Yi Chiang <cychiang(a)chromium.org>: Merge series "Add rt1015 support to CML boards" from Brent Lu <brent.lu(a)intel.com>: Merge series "ASoC: Mediatek: Add support for MT8192 SoC" from Jiaxin Yu <jiaxin.yu(a)mediatek.com>: Merge series "ASoC: topology: Change to resource managed memory" from Amadeusz Sławiński<amadeuszx.slawinski(a)linux.intel.com>: Merge series "ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: ti: davinci-mcasp: Handle incomplete DT node gracefully" from Peter Ujfalusi <peter.ujfalusi(a)ti.com>: Merge series "ASoC: soc-compress: tidyup STREAM vs COMPRESS" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "Fix vaud18 power leakage of mt6359" from Shane Chien <shane.chien(a)mediatek.com> "Shane.Chien" <shane.chien(a)mediatek.com>: Merge series "Modify documentation and machine driver for SC7180 sound card" from Ajye Huang <ajye.huang(a)gmail.com>: Merge series "Audio Graph Updates" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "Fix PM disable depth imbalance on error" from Zhang Qilong <zhangqilong3(a)huawei.com>: Merge series "ASoC: Fix error handling in wm899x" from Zhang Qilong <zhangqilong3(a)huawei.com>: Merge series "ASoC: soundwire: SDCA device quirk and cleanups" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Support SC7180 sound card variation without headset" from xuyuqing <xuyuqing(a)huaqin.corp-partner.google.com>: Merge series "ASoC: SOF: Kconfig corrections" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: use inclusive language for bclk/fsync/topology" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: pcm512x: Patch series to set fmt from `set_fmt()`" from Kirill Marinushkin <kmarinushkin(a)birdec.com>: Merge series "ASoC: Intel: catpt: Offload fixes and code optimization" from Cezary Rojewski <cezary.rojewski(a)intel.com>: Merge series "ASoC: graph card schema rework" from Rob Herring <robh(a)kernel.org>: Merge series "ASoC: soc-component: add snd_soc_component_xxx()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: codecs: add support for LPASS Codec macros" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: Intel/SOF: extend run-time driver selection to ACPI devices" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: mt6359: Fix regulator_dev_lookup() fails for id "LDO_VAUD18"" from Jiaxin Yu <jiaxin.yu(a)mediatek.com>: Merge series "ASoC: fix !OF compile test warnings" from Krzysztof Kozlowski <krzk(a)kernel.org>: Merge series "ASoC: merge soc_compr_open() rollback and soc_compr_free()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: amd: Minor fixes for error handling" from Takashi Iwai <tiwai(a)suse.de>: Merge series "ASoC: tidyup for jack.h" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: SOF: Intel: fix ICL boot sequence" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge branch 'for-5.10' of https://git.kernel.org/.../broonie/sound into asoc-5.11 Merge tag 'soundwire-for-asoc-5.11' of git://git.kernel.org/.../vkoul/soundwire into asoc-5.11 Merge branch '20201104_yung_chuan_liao_regmap_soundwire_asoc_add_soundwire_sdca_support' (early part) into asoc-5.11 Merge series "ASoC: mediatek: mt8192: support new machine rt1015p_rt5682" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge tag 'auxbus-5.11-rc1' of git://git.kernel.org/.../gregkh/driver-core into asoc-5.11 Merge series "ASoC: Intel: cht_bsw_nau8824: 2 fixes for usage with sof-audio-acpi" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: soc-pcm: trigger cleanup" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: rt1015p: delay 300ms for waiting calibration" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: SOF: Intel: fix to dsp state dump trace levels" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge remote-tracking branch 'asoc/for-5.9' into asoc-linus Merge remote-tracking branch 'asoc/for-5.10' into asoc-linus Merge remote-tracking branch 'asoc/for-5.11' into asoc-next Michał Mirosław (1): ASoC: tegra20-spdif: remove "default m" Oder Chiou (2): ASoC: rt5682: Add a new property for the DMIC clock driving ASoC: rt5682: Add a new property to the device tree. Olivier Moysan (3): dt-bindings: stm32: dfsdm: update audio properties ASoC: dt-bindings: stm32: dfsdm: remove stm32-adfsdm.txt binding ASoC: dt-bindings: stm32: convert sai to json-schema Paul Cercueil (3): ASoC: codecs/jz47xx: Use regmap_{set,clear}_bits ASoC: codecs/jz4770: Don't change cap-couple setting in HP PMU/PMD ASoC: codecs/jz4770: Add DAPM widget to set HP out to cap-less mode Peter Ujfalusi (4): ASoC: ti: davinci-mcasp: Use platform_get_irq_byname_optional ASoC: ti: davinci-mcasp: Remove legacy dma_request parsing ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling ASoC: ti: davinci-mcasp: Handle missing required DT properties Pierre-Louis Bossart (31): ASoC: SOF: control: remove const in sizeof() ASoC: SOF: topology: remove const in sizeof() ASoC: SOF: sof-audio: remove goto used for force-nocodec support ASoC: Intel: sof_sdw: add quirk for new TigerLake-SDCA device ASoC: codecs: rt*.c: remove useless pointer cast ASoC: rt711: remove warnings ASoC: codecs: max98373-sdw: align regmap use with other codecs ASoC: SOF: Intel: fix Kconfig dependency for SND_INTEL_DSP_CONFIG ASoC: topology: use inclusive language for bclk and fsync ASoC: SOF: use inclusive language for bclk and fsync ASoC: Intel: atom: use inclusive language for SSP bclk/fsync ASoC: Intel: keembay: use inclusive language for bclk and fsync ASoC: Intel: broadwell: add missing pm_ops ASoC: Intel: bdw-rt5677: add missing pm_ops ALSA: hda: intel-dsp-config: add helper for ACPI DSP driver selection ASoC: soc-acpi: add helper to identify parent driver. ASoC: Intel: boards: byt/cht: set card and driver name at run time ASoC: Intel: byt/cht: set pm ops dynamically ASoC: SOF: acpi: add dynamic selection of DSP driver ASoC: Intel: Atom: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and Atom/SST drivers ALSA: hda: intel-dsp-config: add Broadwell ACPI DSP driver selection ASoC: Intel: broadwell: set card and driver name dynamically ASoC: Intel: catpt: add dynamic selection of DSP driver ALSA: hda: intel-dsp-config: ignore dsp_driver parameter for PCI legacy devices ASoC: SOF: Intel: allow for coexistence between SOF and catpt drivers soundwire: SDCA: detect sdca_cascade interrupt soundwire: SDCA: add helper macro to access controls regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ ASoC: SOF: control: fix cppcheck warning in snd_sof_volume_info() ASoC: SOF: imx: update kernel-doc description Piotr Maziarz (1): ASoC: Intel: catpt: select WANT_DEV_COREDUMP Randy Dunlap (3): ASoC: SOF: imx: fix Kconfig punctuation ASoC: SOF: Kconfig: fix Kconfig punctuation and wording ASoC: SOF: Intel: fix Kconfig punctuation and wording Ranjani Sridharan (6): ASoC: pcm: call snd_soc_dapm_stream_stop() in soc_pcm_hw_clean ASoC: SOF: nocodec: modify DAI link definitions ASoC: pcm: send DAPM_STREAM_STOP event in dpcm_fe_dai_shutdown ASoC: SOF: Intel: hda: remove duplicated status dump ASoC: SOF: modify the SOF_DBG flags ASoC: SOF: Intel: hda: fix the condition passed to sof_dev_dbg_or_err Richard Fitzgerald (1): ASoC: wm5102: Use get_unaligned_be16() for dac_comp_coeff Rob Herring (3): ASoC: dt-bindings: marvell, mmp-sspa: Use audio-graph-port schema ASoC: dt-bindings: Refine 'frame-master' and 'bitclock-master' type ASoC: dt-bindings: renesas, rsnd: Fix duplicate 'allOf' entries Sameer Pujar (10): ASoC: soc-core: Fix component name_prefix parsing ASoC: soc-pcm: Get all BEs along DAPM path ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: audio-graph: Support empty Codec endpoint ASoC: audio-graph: Expose new members for asoc_simple_priv ASoC: audio-graph: Expose helpers from audio graph ASoC: audio-graph-card: Refactor schema dt-bindings: tegra: Convert HDA doc to json-schema dt-bindings: tegra: Add missing HDA properties Samuel Holland (18): ASoC: sun8i-codec: Prepare to extend the DAI driver ASoC: sun8i-codec: Program DAI format before clock inversion ASoC: sun8i-codec: Enable all supported clock inversions ASoC: sun8i-codec: Use the provided word size ASoC: sun8i-codec: Round up the LRCK divisor ASoC: sun8i-codec: Correct the BCLK divisor calculation ASoC: sun8i-codec: Support the TDM slot binding ASoC: sun8i-codec: Enforce symmetric DAI parameters ASoC: sun8i-codec: Enable all supported sample rates ASoC: sun8i-codec: Automatically set the system sample rate ASoC: sun8i-codec: Constrain to compatible sample rates ASoC: sun8i-codec: Protect the clock rate while streams are open ASoC: sun8i-codec: Require an exact BCLK divisor match ASoC: sun8i-codec: Enable all supported PCM formats ASoC: sun8i-codec: Generalize AIF clock control ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes ASoC: sun4i-i2s: Fix setting of FIFO modes Sathyanarayana Nujella (1): ASoC: Intel: Boards: tgl_max98373: update TDM slot_width Shane.Chien (3): ASoC: Fix vaud18 power leakage of mt6359 dt-bindings: mediatek: mt6359: Add new property for mt6359 ASoC: Remove mt6359_platform_driver_remove Shengjiu Wang (13): ASoC: dt-bindings: fsl_spdif: Add new compatible string for i.MX8QM ASoC: fsl_spdif: Add support for i.MX8QM platform ASoC: fsl_spdif: Add support for higher sample rates ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module ASoC: fsl_aud2htx: Add aud2htx module driver ASoC: fsl_aud2htx: Remove dev_err() usage after platform_get_irq() ASoC: fsl_sai: Correct the clock source for mclk0 ASoC: hdmi-codec: Add RX support ASoC: fsl: Fix config name of CONFIG_ARCH_MXC ASoC: fsl-asoc-card: Add support for si476x codec ASoC: bindings: fsl-asoc-card: add compatible string for si476x codec ASoC: dt-bindings: imx-hdmi: Add binding doc for hdmi machine driver ASoC: fsl: Add imx-hdmi machine driver Shuming Fan (1): ASoC: rt1015: support TDM slot configuration Srinivas Kandagatla (17): ASoC: qcom: dt-bindings: Add SM8250 sound card bindings ASoC: qcom: sm8250: add sound card qrb5165-rb5 support ASoC: qcom: qdsp6: make use of devm_of_platform_populate ASoC: qcom: sm8250: Fix array out of bounds access ASoC: qcom: dt-bindings: sm8250: update compatibles ASoC: qcom: sm8250: update compatible with new bindings ASoC: qcom: lpass-cpu: fix warning on symbol scope ASoC: q6afe-clocks: fix warning on symbol scope ASoC: qcom: dt-bindings: add bindings for lpass wsa macro codec ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro ASoC: codecs: lpass-wsa-macro: add dapm widgets and route ASoC: qcom: dt-bindings: add bindings for lpass va macro codec ASoC: codecs: lpass-va-macro: Add support to VA Macro ASoC: codecs: lpass-va-macro: add dapm widgets and routes ASoC: codecs: lpass-va-macro: add missing MODULE_DEVICE_TABLE soundwire: Fix DEBUG_LOCKS_WARN_ON for uninitialized attribute ASoC: q6afe-clocks: Add missing parent clock rate Sudip Mukherjee (1): ASoC: mediatek: mt8192: Fix build failure Takashi Iwai (3): ASoC: amd: Downgrade print level for invalid ACP mode ASoC: amd: Return -ENODEV for non-existing ACPI call ASoC: cx2072x: Fix doubly definitions of Playback and Capture streams Tom Rix (5): ASoC: bcm2835-i2s: remove unneeded semicolon ASoC: Intel: remove unneeded semicolon ASoC: TSCS454: remove unneeded semicolon ASoC: tegra: remove unneeded semicolon ASoC: TSCS42xx: remove unneeded semicolon Tzung-Bi Shih (9): ASoC: mediatek: mt8192: rename common symbols ASoC: mediatek: mt8192: extract rt1015_rt5682 specific DAI link ASoC: mediatek: mt8192: move rt1015_rt5682 specific data ASoC: dt-bindings: mt8192-mt6359: fix typo in the example ASoC: dt-bindings: mt8192-mt6359: add new compatible for using rt1015p ASoC: mediatek: mt8192: support rt1015p_rt5682 ASoC: rt1015p: move SDB control from trigger to DAPM ASoC: rt1015p: delay 300ms after SDB pulling high for calibration ASoC: mediatek: mt8183: add PM ops to machine drivers V Sujith Kumar Reddy (2): ASoC: qcom: lpass-sc7180: Add 32 bit format support for capture ASoC: qcom: sc7180: Register shutdown handler for lpass platform Vijendar Mukunda (1): ASoC: amd: support other audio modes for raven Viorel Suman (4): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR ASoC: fsl_xcvr: fix break condition ASoC: fsl_xcvr: fix potential resource leak Xu Wang (3): ASoC: ti: davinci-evm: Remove redundant null check before clk_disable_unprepare ASoC: samsung: i2s: Remove redundant null check before clk_disable_unprepare ASoC: pxa: pxa-ssp: Remove redundant null check before clk_prepare_enable/clk_disable_unprepare Zhang Qilong (4): ASoC: ti: davinci-mcasp: remove always zero of davinci_mcasp_get_dt_params ASoC: wm8994: Fix PM disable depth imbalance on error ASoC: wm8998: Fix PM disable depth imbalance on error ASoC: arizona: Fix a wrong free in wm8997_probe Zou Wei (2): ASoC: mediatek: mt8192: Make some symbols static ASoC: samsung: fix platform_no_drv_owner.cocci warnings xuyuqing (3): ASoC: google: dt-bindings: add new compatible for sc7180-coachz ASoC: qcom: sc7180: Modify machine driver for sound card ASoC: qcom: sc7180: fix 32 bit format for adau7002 CREDITS | 9 + Documentation/ABI/README | 10 +- Documentation/ABI/obsolete/sysfs-class-dax | 8 +- .../ABI/obsolete/sysfs-driver-hid-roccat-pyra | 3 + Documentation/ABI/obsolete/sysfs-gpio | 2 + Documentation/ABI/removed/devfs | 1 + Documentation/ABI/removed/raw1394 | 1 + Documentation/ABI/removed/sysfs-class-rfkill | 2 +- Documentation/ABI/removed/video1394 | 1 + Documentation/ABI/stable/firewire-cdev | 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8 +- samples/mic/mpssd/.gitignore | 2 - samples/mic/mpssd/Makefile | 28 - samples/mic/mpssd/micctrl | 162 - samples/mic/mpssd/mpss | 189 -- samples/mic/mpssd/mpssd.c | 1815 ------------ samples/mic/mpssd/mpssd.h | 89 - samples/mic/mpssd/sysfs.c | 91 - scripts/get_abi.pl | 388 ++- scripts/kernel-doc | 21 +- scripts/mod/devicetable-offsets.c | 3 + scripts/mod/file2alias.c | 8 + security/integrity/ima/ima.h | 2 +- sound/core/control.c | 4 +- sound/core/pcm_dmaengine.c | 3 +- sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 4 +- sound/hda/ext/hdac_ext_controller.c | 2 + sound/hda/intel-dsp-config.c | 111 + sound/pci/hda/hda_codec.c | 45 +- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 63 +- sound/pci/hda/patch_realtek.c | 67 +- sound/soc/adi/Kconfig | 1 - sound/soc/amd/acp-da7219-max98357a.c | 9 +- sound/soc/amd/raven/pci-acp3x.c | 9 +- sound/soc/amd/renoir/rn-pci-acp3x.c | 34 +- sound/soc/atmel/Kconfig | 1 + sound/soc/atmel/atmel-i2s.c | 4 +- sound/soc/bcm/bcm2835-i2s.c | 4 +- sound/soc/codecs/Kconfig | 63 +- sound/soc/codecs/Makefile | 19 + sound/soc/codecs/adau1372-i2c.c | 40 + sound/soc/codecs/adau1372-spi.c | 58 + sound/soc/codecs/adau1372.c | 1062 +++++++ sound/soc/codecs/adau1372.h | 21 + sound/soc/codecs/adau1977.c | 9 +- sound/soc/codecs/adav80x.c | 1 + sound/soc/codecs/ak4118.c | 2 + sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/alc5623.c | 2 + sound/soc/codecs/alc5632.c | 2 + sound/soc/codecs/arizona.c | 1 + sound/soc/codecs/bd28623.c | 2 +- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/cs42l52.c | 1 + sound/soc/codecs/cs42l56.c | 1 + sound/soc/codecs/cs47l92.c | 1 + sound/soc/codecs/cx2072x.c | 4 +- sound/soc/codecs/da7218.c | 2 + sound/soc/codecs/da7219.c | 2 + sound/soc/codecs/da9055.c | 2 + sound/soc/codecs/es7134.c | 4 +- sound/soc/codecs/es7241.c | 2 +- sound/soc/codecs/es8316.c | 2 + sound/soc/codecs/gtm601.c | 2 +- sound/soc/codecs/hdmi-codec.c | 43 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/jz4725b.c | 26 +- sound/soc/codecs/jz4740.c | 20 +- sound/soc/codecs/jz4770.c | 109 +- sound/soc/codecs/lpass-va-macro.c | 1497 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 2464 +++++++++++++++ sound/soc/codecs/lpass-wsa-macro.h | 17 + sound/soc/codecs/madera.c | 4 +- sound/soc/codecs/max98090.c | 2 + sound/soc/codecs/max98095.c | 2 + sound/soc/codecs/max98371.c | 3 +- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/max98390.c | 2 + sound/soc/codecs/max9867.c | 2 + sound/soc/codecs/max98925.c | 3 +- sound/soc/codecs/max98926.c | 3 +- sound/soc/codecs/mt6359.c | 152 +- sound/soc/codecs/mt6359.h | 15 +- sound/soc/codecs/nau8315.c | 166 ++ sound/soc/codecs/pcm1789-i2c.c | 2 + sound/soc/codecs/pcm179x-i2c.c | 2 + sound/soc/codecs/pcm512x.c | 134 +- sound/soc/codecs/rk3328_codec.c | 2 +- sound/soc/codecs/rt1015.c | 211 +- sound/soc/codecs/rt1015.h | 42 + sound/soc/codecs/rt1015p.c | 69 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5660.c | 2 + sound/soc/codecs/rt5682-i2c.c | 5 + sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt5682.c | 3 + sound/soc/codecs/rt5682.h | 14 + sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdw.c | 15 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdca-sdw.c | 278 ++ sound/soc/codecs/rt715-sdca-sdw.h | 170 ++ sound/soc/codecs/rt715-sdca.c | 936 ++++++ sound/soc/codecs/rt715-sdca.h | 124 + sound/soc/codecs/rt715.c | 2 +- sound/soc/codecs/rt715.h | 1 - sound/soc/codecs/simple-mux.c | 124 + sound/soc/codecs/tas2562.c | 2 + sound/soc/codecs/tas571x.c | 4 +- sound/soc/codecs/tlv320adcx140.c | 2 + sound/soc/codecs/tlv320aic23-i2c.c | 2 + sound/soc/codecs/ts3a227e.c | 2 + sound/soc/codecs/tscs42xx.c | 6 +- sound/soc/codecs/tscs454.c | 8 +- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm8350.c | 3 +- sound/soc/codecs/wm8962.c | 1 + sound/soc/codecs/wm8994.c | 6 +- sound/soc/codecs/wm8997.c | 2 + sound/soc/codecs/wm8998.c | 4 +- sound/soc/codecs/wm_adsp.c | 5 +- sound/soc/fsl/Kconfig | 77 +- sound/soc/fsl/Makefile | 16 +- sound/soc/fsl/fsl-asoc-card.c | 13 + sound/soc/fsl/fsl_aud2htx.c | 308 ++ sound/soc/fsl/fsl_aud2htx.h | 67 + sound/soc/fsl/fsl_audmix.c | 13 +- sound/soc/fsl/fsl_sai.c | 20 +- sound/soc/fsl/fsl_sai.h | 1 + sound/soc/fsl/fsl_spdif.c | 73 +- sound/soc/fsl/fsl_spdif.h | 9 +- sound/soc/fsl/fsl_xcvr.c | 1360 +++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++ sound/soc/fsl/imx-audmux.c | 28 +- sound/soc/fsl/imx-hdmi.c | 236 ++ sound/soc/fsl/imx-mc13783.c | 156 - sound/soc/fsl/imx-ssi.c | 651 ---- sound/soc/fsl/mx27vis-aic32x4.c | 214 -- sound/soc/fsl/phycore-ac97.c | 121 - sound/soc/fsl/wm1133-ev1.c | 289 -- sound/soc/generic/Kconfig | 2 +- sound/soc/generic/audio-graph-card.c | 175 +- sound/soc/intel/Kconfig | 5 +- sound/soc/intel/atom/sst-atom-controls.c | 12 +- sound/soc/intel/atom/sst-atom-controls.h | 4 +- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 8 + sound/soc/intel/boards/Kconfig | 3 +- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/bdw-rt5650.c | 17 +- sound/soc/intel/boards/bdw-rt5677.c | 18 +- sound/soc/intel/boards/broadwell.c | 20 +- sound/soc/intel/boards/bytcht_cx2072x.c | 27 +- sound/soc/intel/boards/bytcht_da7213.c | 27 +- sound/soc/intel/boards/bytcht_es8316.c | 29 +- sound/soc/intel/boards/bytcr_rt5640.c | 42 +- sound/soc/intel/boards/bytcr_rt5651.c | 29 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 29 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 39 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 38 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 29 +- sound/soc/intel/boards/sof_maxim_common.c | 4 +- sound/soc/intel/boards/sof_realtek_common.c | 138 + sound/soc/intel/boards/sof_realtek_common.h | 24 + sound/soc/intel/boards/sof_rt5682.c | 87 +- sound/soc/intel/boards/sof_sdw.c | 10 + sound/soc/intel/catpt/core.h | 10 +- sound/soc/intel/catpt/device.c | 30 +- sound/soc/intel/catpt/dsp.c | 56 +- sound/soc/intel/catpt/loader.c | 2 +- sound/soc/intel/catpt/pcm.c | 106 +- sound/soc/intel/common/Makefile | 2 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 52 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 13 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 17 +- sound/soc/intel/keembay/kmb_platform.c | 22 +- sound/soc/intel/keembay/kmb_platform.h | 8 +- sound/soc/intel/skylake/skl-topology.c | 9 +- sound/soc/jz4740/jz4740-i2s.c | 9 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mediatek/Kconfig | 25 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 13 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-clk.c | 1 - sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 + sound/soc/mediatek/mt8192/Makefile | 16 + sound/soc/mediatek/mt8192/mt8192-afe-clk.c | 669 +++++ sound/soc/mediatek/mt8192/mt8192-afe-clk.h | 244 ++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 170 ++ sound/soc/mediatek/mt8192/mt8192-afe-control.c | 163 + sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 308 ++ sound/soc/mediatek/mt8192/mt8192-afe-gpio.h | 19 + sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2389 +++++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 1471 +++++++++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2110 +++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 409 +++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 778 +++++ sound/soc/mediatek/mt8192/mt8192-interconnection.h | 65 + .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 1137 +++++++ sound/soc/mediatek/mt8192/mt8192-reg.h | 3131 ++++++++++++++++++++ sound/soc/meson/Kconfig | 4 +- sound/soc/meson/t9015.c | 2 +- sound/soc/pxa/Kconfig | 14 +- sound/soc/pxa/mmp-sspa.c | 4 + sound/soc/pxa/pxa-ssp.c | 6 +- sound/soc/qcom/Kconfig | 25 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/common.c | 13 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-cpu.c | 12 +- sound/soc/qcom/lpass-hdmi.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 10 +- sound/soc/qcom/lpass.h | 1 + sound/soc/qcom/qdsp6/q6adm.c | 10 +- sound/soc/qcom/qdsp6/q6afe-clocks.c | 3 +- sound/soc/qcom/qdsp6/q6afe.c | 10 +- sound/soc/qcom/qdsp6/q6asm.c | 10 +- sound/soc/qcom/sc7180.c | 391 +++ sound/soc/qcom/sm8250.c | 229 ++ sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s.c | 11 +- sound/soc/samsung/midas_wm1811.c | 1 - sound/soc/samsung/smdk_wm8994.c | 4 +- sound/soc/samsung/snow.c | 2 +- sound/soc/sh/Kconfig | 1 + sound/soc/soc-component.c | 318 +- sound/soc/soc-compress.c | 397 +-- sound/soc/soc-core.c | 90 +- sound/soc/soc-dai.c | 70 +- sound/soc/soc-dapm.c | 8 +- sound/soc/soc-link.c | 53 +- sound/soc/soc-pcm.c | 217 +- sound/soc/soc-topology.c | 337 +-- sound/soc/sof/Kconfig | 42 +- sound/soc/sof/control.c | 42 +- sound/soc/sof/debug.c | 119 +- sound/soc/sof/imx/Kconfig | 10 +- sound/soc/sof/imx/imx-common.c | 2 + sound/soc/sof/intel/Kconfig | 110 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/apl.c | 3 + sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/intel/cnl.c | 19 +- sound/soc/sof/intel/ext_manifest.h | 35 + sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-loader.c | 119 +- sound/soc/sof/intel/hda-pcm.c | 7 + sound/soc/sof/intel/hda.c | 10 +- sound/soc/sof/intel/hda.h | 12 + sound/soc/sof/intel/icl.c | 145 + sound/soc/sof/intel/intel-ipc.c | 7 + sound/soc/sof/intel/tgl.c | 19 + sound/soc/sof/ipc.c | 9 + sound/soc/sof/loader.c | 62 +- sound/soc/sof/nocodec.c | 18 +- sound/soc/sof/ops.c | 2 +- sound/soc/sof/ops.h | 14 +- sound/soc/sof/pcm.c | 14 +- sound/soc/sof/sof-acpi-dev.c | 14 +- sound/soc/sof/sof-audio.c | 12 +- sound/soc/sof/sof-audio.h | 5 + sound/soc/sof/sof-pci-dev.c | 24 +- sound/soc/sof/sof-priv.h | 22 +- sound/soc/sof/topology.c | 33 +- sound/soc/sof/trace.c | 224 ++ sound/soc/stm/Kconfig | 1 + sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 388 ++- sound/soc/sunxi/sun8i-codec.c | 834 +++++- sound/soc/tegra/Kconfig | 1 - sound/soc/tegra/tegra186_dspk.c | 6 +- sound/soc/tegra/tegra210_dmic.c | 6 +- sound/soc/tegra/tegra210_i2s.c | 6 +- sound/soc/tegra/tegra_alc5632.c | 4 +- sound/soc/tegra/tegra_max98090.c | 8 +- sound/soc/tegra/tegra_rt5640.c | 8 +- sound/soc/tegra/tegra_rt5677.c | 4 +- sound/soc/tegra/tegra_sgtl5000.c | 4 +- sound/soc/tegra/tegra_wm8753.c | 8 +- sound/soc/tegra/tegra_wm8903.c | 8 +- sound/soc/tegra/tegra_wm9712.c | 4 +- sound/soc/tegra/trimslice.c | 8 +- sound/soc/ti/Kconfig | 9 +- sound/soc/ti/davinci-evm.c | 3 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 298 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- sound/usb/pcm.c | 6 + sound/usb/quirks.c | 1 + tools/arch/arm64/include/uapi/asm/kvm.h | 25 + tools/arch/s390/include/uapi/asm/sie.h | 2 +- tools/arch/x86/include/asm/cpufeatures.h | 6 +- tools/arch/x86/include/asm/disabled-features.h | 9 +- tools/arch/x86/include/asm/msr-index.h | 10 + tools/arch/x86/include/asm/required-features.h | 2 +- tools/arch/x86/include/uapi/asm/kvm.h | 20 + tools/arch/x86/include/uapi/asm/svm.h | 13 + tools/build/feature/test-all.c | 1 - tools/include/linux/compiler-gcc.h | 12 - tools/include/linux/compiler.h | 3 - tools/include/uapi/asm-generic/unistd.h | 4 +- tools/include/uapi/drm/i915_drm.h | 59 +- tools/include/uapi/linux/fscrypt.h | 6 +- tools/include/uapi/linux/kvm.h | 19 + tools/include/uapi/linux/mman.h | 1 + tools/include/uapi/linux/mount.h | 1 + tools/include/uapi/linux/perf_event.h | 2 +- tools/include/uapi/linux/prctl.h | 9 + tools/include/uapi/linux/vhost.h | 4 + tools/perf/Makefile.config | 1 + 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drivers/misc/mic/scif/scif_nodeqp.h delete mode 100644 drivers/misc/mic/scif/scif_peer_bus.c delete mode 100644 drivers/misc/mic/scif/scif_peer_bus.h delete mode 100644 drivers/misc/mic/scif/scif_ports.c delete mode 100644 drivers/misc/mic/scif/scif_rb.c delete mode 100644 drivers/misc/mic/scif/scif_rb.h delete mode 100644 drivers/misc/mic/scif/scif_rma.c delete mode 100644 drivers/misc/mic/scif/scif_rma.h delete mode 100644 drivers/misc/mic/scif/scif_rma_list.c delete mode 100644 drivers/misc/mic/scif/scif_rma_list.h delete mode 100644 drivers/misc/mic/vop/Makefile delete mode 100644 drivers/misc/mic/vop/vop_debugfs.c delete mode 100644 drivers/misc/mic/vop/vop_main.c delete mode 100644 drivers/misc/mic/vop/vop_main.h delete mode 100644 drivers/misc/mic/vop/vop_vringh.c create mode 100644 include/dt-bindings/sound/adi,adau1977.h create mode 100644 include/linux/auxiliary_bus.h delete mode 100644 include/linux/mic_bus.h delete mode 100644 include/linux/platform_data/adau1977.h delete mode 100644 include/linux/scif.h create mode 100644 include/sound/graph_card.h create mode 100644 include/sound/soc-jack.h create mode 100644 include/sound/sof/debug.h delete mode 100644 include/uapi/linux/mic_common.h delete mode 100644 include/uapi/linux/mic_ioctl.h delete mode 100644 samples/mic/mpssd/.gitignore delete mode 100644 samples/mic/mpssd/Makefile delete mode 100755 samples/mic/mpssd/micctrl delete mode 100755 samples/mic/mpssd/mpss delete mode 100644 samples/mic/mpssd/mpssd.c delete mode 100644 samples/mic/mpssd/mpssd.h delete mode 100644 samples/mic/mpssd/sysfs.c create mode 100644 sound/soc/codecs/adau1372-i2c.c create mode 100644 sound/soc/codecs/adau1372-spi.c create mode 100644 sound/soc/codecs/adau1372.c create mode 100644 sound/soc/codecs/adau1372.h create mode 100644 sound/soc/codecs/lpass-va-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.h create mode 100644 sound/soc/codecs/nau8315.c create mode 100644 sound/soc/codecs/rt715-sdca-sdw.c create mode 100644 sound/soc/codecs/rt715-sdca-sdw.h create mode 100644 sound/soc/codecs/rt715-sdca.c create mode 100644 sound/soc/codecs/rt715-sdca.h create mode 100644 sound/soc/codecs/simple-mux.c create mode 100644 sound/soc/fsl/fsl_aud2htx.c create mode 100644 sound/soc/fsl/fsl_aud2htx.h create mode 100644 sound/soc/fsl/fsl_xcvr.c create mode 100644 sound/soc/fsl/fsl_xcvr.h create mode 100644 sound/soc/fsl/imx-hdmi.c delete mode 100644 sound/soc/fsl/imx-mc13783.c delete mode 100644 sound/soc/fsl/imx-ssi.c delete mode 100644 sound/soc/fsl/mx27vis-aic32x4.c delete mode 100644 sound/soc/fsl/phycore-ac97.c delete mode 100644 sound/soc/fsl/wm1133-ev1.c create mode 100644 sound/soc/intel/boards/sof_realtek_common.c create mode 100644 sound/soc/intel/boards/sof_realtek_common.h create mode 100644 sound/soc/intel/common/soc-acpi-intel-adl-match.c create mode 100644 sound/soc/mediatek/mt8192/Makefile create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-common.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-control.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-adda.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-interconnection.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-reg.h create mode 100644 sound/soc/qcom/sc7180.c create mode 100644 sound/soc/qcom/sm8250.c create mode 100644 sound/soc/sof/intel/ext_manifest.h create mode 100644 sound/soc/sof/intel/icl.c create mode 100644 tools/testing/selftests/kvm/x86_64/vmx_apic_access_test.c

2 1

[RFC][PATCH 0/2] design a way to change audio Jack state by software
by Hui Wang 14 Dec '20

14 Dec '20

After we change sth in the userspace audio stack like alsa-ucm or pulseaudio, we want to perform remote audio auto test to verify if the change introduce the regression or not, some of the tests are about the defaut_sink/default_source or active_port switching, this needs the audio jack state to be changed to trigger the userspace's audio device switching. So far, there is no software ways to change the audio jack state, this block the auto test. My design is adding a sysfs interface for each sound card if the card has audio jack, then users could echo different values to sysfs to change the jack state (Phantom jack is not controlled by injection). And once the users enable the jack injection via sysfs, this jack's state will not be controlled by hw events anymore until users disable the jack injection. Of course, this could not 100% simulate the plugin or plugout triggered by hw events, with the hw triggered plugin or plugout, the audio driver will set codec or does sth else, so the software injection is just changing the jack state and notify the userspace, it is just for testing userspace part. Here is an example to change jack state via sysfs: After booting up: /* cd to the jack injection folder for sound card0 in the sysfs */ $cd /sys/devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0/jack /* check file nodes in this folder */ $ls jackin_inject sw_inject_enable /* check all jack's software injection enable status, all disabled now */ $ cat sw_inject_enable Jack: Mic 0 Jack: Headphone 0 Jack: HDMI/DP,pcm=3 0 Jack: HDMI/DP,pcm=4 0 Jack: HDMI/DP,pcm=5 0 /* enable software injection for Jack Headphone */ $ sudo sh -c "echo Headphone 1 > sw_inject_enable" /* check all jack's software injection enable status again, now Headphone is enabled */ $ cat sw_inject_enable Jack: Mic 0 Jack: Headphone 1 Jack: HDMI/DP,pcm=3 0 Jack: HDMI/DP,pcm=4 0 Jack: HDMI/DP,pcm=5 0 /* trigger plugin to Jack Headphone */ $sudo sh -c "echo Headphone 1 > jackin_inject" /* check if Jack Headphone is plugged in */ $ sudo amixer contents | grep "Headphone Jack" -3 numid=30,iface=CARD,name='HDMI/DP,pcm=5 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=17,iface=CARD,name='Headphone Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=14,iface=CARD,name='Mic Jack' /* trigger plugout to Jack Headphone */ $ sudo sh -c "echo Headphone 0 > jackin_inject" /* check if Jack Headphone is plugged out */ $ sudo amixer contents | grep "Headphone Jack" -3 numid=30,iface=CARD,name='HDMI/DP,pcm=5 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=17,iface=CARD,name='Headphone Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=14,iface=CARD,name='Mic Jack' /* disable Jack Headphone software injection, this will return the control to non-injection ways */ $ sudo sh -c "echo Headphone 0 > sw_inject_enable" /* check if the Jack Headphone software injection is disabled, it is disabled now */ $ cat sw_inject_enable Jack: Mic 0 Jack: Headphone 0 Jack: HDMI/DP,pcm=3 0 Jack: HDMI/DP,pcm=4 0 Jack: HDMI/DP,pcm=5 0 Hui Wang (2): alsa: jack: expand snd_jack_report parameter for jack sw_inject alsa: jack: adding support for software jack in or out injection include/sound/core.h | 1 + include/sound/jack.h | 5 +- sound/core/jack.c | 129 +++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.c | 6 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 2 +- sound/x86/intel_hdmi_audio.c | 4 +- 8 files changed, 140 insertions(+), 11 deletions(-) -- 2.25.1

4 7

[PATCH v3 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 14 Dec '20

14 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

1 0

[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 14 Dec '20

14 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0

[PATCH alsa-lib 1/3] chtnau8824: Fix mono speaker config not working
by Hans de Goede 14 Dec '20

14 Dec '20

Fix the Speaker define / var being changed from "Speaker" to "MonoSpeaker" not having any effect because the variable gets expanded before the if making the change is interpreted. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/chtnau8824/HiFi.conf | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf index 3ccd6b1..d7a5f63 100644 --- a/ucm2/chtnau8824/HiFi.conf +++ b/ucm2/chtnau8824/HiFi.conf @@ -34,8 +34,15 @@ SectionVerb { } } -Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" -Include.hp.File "/codecs/nau8824/HeadPhones.conf" +# The includes using $vars in there path must be conditional otherwise the +# $var gets expanded before the other If-s above can change the vars. +If.cfg-includes { + Condition { Type String Empty "" } + True { + Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" + Include.hp.File "/codecs/nau8824/HeadPhones.conf" -Include.mic.File "/codecs/nau8824/InternalMic.conf" -Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" + Include.mic.File "/codecs/nau8824/InternalMic.conf" + Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" + } +} -- 2.28.0

2 4

[RFC PATCH 0/4] Enable DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 14 Dec '20

14 Dec '20

v1: Initial patch version, to enable DMA mode on Intel Keembay platform by exposing some dmaengine api to work around DMA limitations at the client driver level. This patchset suggests an ALSA-only quirk, having other more generic means to deal with this limitation would be fine - we just wanted to have a discussion on preferred directions. The IPs used are not Intel-specific so sooner or later someone else will have similar limitations to work-around. Michael Sit Wei Hong (4): dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: soc-generic-dmaengine-pcm: Add custom prepare and submit function ASoC: dmaengine_pcm: expose functions to header file for custom functions ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ include/sound/dmaengine_pcm.h | 21 ++ sound/core/pcm_dmaengine.c | 46 ++-- sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 202 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + sound/soc/soc-generic-dmaengine-pcm.c | 8 +- 7 files changed, 267 insertions(+), 35 deletions(-) -- 2.17.1

8 22

Problem with synthesized output
by Spencer Kohan 13 Dec '20

13 Dec '20

Hello, I hope I am posting this to the correct place. I'm trying to use a Raspberry Pi as a wifi speaker, and as a first step I'm trying to just play some sound out of a buffer through the attached amp. I'm using an AMP2 with a RPi 4, and I'm getting some strange behavior when I try to play any "sine" output. So for instance, if I use `speaker-test`, I am able to play the wav test and the pink-noise test just fine, but when I try to play the sine test, it plays for like one quick instant, and then the program blocks. What's strange is I get the exact same behavior when I use other methods to try to play a sine output over ASLA. For instance, if I use sox: play -n synth sine 1000 I get the same output: one quick blip of the tone, and then the program blocks. The output to the console looks like this: play WARN alsa: can't encode 0-bit Unknown or not applicable File Size: 0 Encoding: n/a Channels: 1 @ 32-bit Samplerate: 48000Hz Replaygain: off Duration: unknown In:0.00% 00:00:00.00 [00:00:00.00] Out:0 [ | ] Clip:0 And then I get the exact same behavior when I try to do the `sio_sine` example from [libsoundio](https://github.com/andrewrk/libsoundio), with this output: Backend: ALSA ALSA lib pcm_dsnoop.c:575:(snd_pcm_dsnoop_open) The dsnoop plugin supports only capture stream Output device: snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0: Default Audio Device Software latency: 0.008000 'p\n' - pause 'u\n' - unpause 'P\n' - pause from within callback 'c\n' - clear buffer 'q\n' - quit unable to set channel layout: incompatible device Also interesting, when I try the [rsoundio]( https://github.com/klingtnet/rsoundio) sine example, it reports an underflow. My output for aplay looks like this: $ aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=sndrpihifiberry snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Default Audio Device sysdefault:CARD=sndrpihifiberry snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Default Audio Device dmix:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct sample mixing device dsnoop:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct sample snooping device hw:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct hardware device without any conversions plughw:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Hardware device with all software conversions It's just strange because the pre-recorded sample and the pink noise version work just fine. How would I even go about trying to debug this?

1 0

Bug in the ALSA driver 'snd_usb_audio'
by GitHub issues - opened 12 Dec '20

12 Dec '20

alsa-project/alsa-lib issue #105 was opened from rombru: Hello, I'm not an expert in linux systems. I will try to give a maximum of information. I've bought Bowers & Wilkins PX7 headphones. They doesn't work using USB on Ubuntu. Using Ubuntu and bluetooth, or audio jack, the headphones work well. Using Windows 10 and USB, the headphones work well. I d'like too use them with USB on Ubuntu because that recharge them in parallel. Problems while the headphons are connected : On Chrome : The videos are played too fast and there's no sound (Youtube & Twitch). On Firefox : The videos are played too fast and there's a crackling sound (Youtube & Twitch). On VLC : No sound I've tried with Ubuntu 20.04 & 18.04. They are set as default audio device I've played with PulseAudio and alsamixer (reinstall, mute/unmute, ...) The headphones are set as fallback in PulseAudio. Trying to find the issue, i've make some test found on Google : I've added pcie_aspm=off in GRUB_CMDLINE_LINUX_DEFAULT Setting devices as default in /etc/asound.conf --- The ALSA part I think the problem come from ALSA, i've found the errors : `Failed to load module "module-alsa-card` followed few lines later by `Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers.` Output of `journalctl --user -u pulseaudio` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Output of `pulseaudio -vvvvv` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Output of `pacmd ls` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Please, tell me if you need something more to analyse the problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/105 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 5.9 06/23] ASoC: rt5682: change SAR voltage threshold
by Sasha Levin 12 Dec '20

12 Dec '20

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit aa4cb898b80a28a610e26d1513e6dd42d995c225 ] To fix errors in some 4 poles headset detection cases, this patch adjusts the voltage threshold for mic detection. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20201126092759.9427-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index a4713bd6508d5..e260eddb51d96 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,6 +43,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, + {RT5682_SAR_IL_CMD_3, 0x8365}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.27.0

1 0

[PATCH 1/2] ALSA: hda/ca0132 - Add 8051 PLL write helper functions.
by Connor McAdams 12 Dec '20

12 Dec '20

Add helper functions for the 8051 PLL PMU write verbs. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 130 ++++++++++++++--------------------- 1 file changed, 50 insertions(+), 80 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e9019e840a4d..9b6b7601a881 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1949,6 +1949,14 @@ static unsigned int chipio_8051_get_data(struct hda_codec *codec) VENDOR_CHIPIO_8051_DATA_READ, 0); } +/* PLL_PMU writes share the lower address register of the 8051 exram writes. */ +static void chipio_8051_set_data_pll(struct hda_codec *codec, unsigned int data) +{ + /* 8-bits of data. */ + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, data & 0xff); +} + static void chipio_8051_write_exram(struct hda_codec *codec, unsigned int addr, unsigned int data) { @@ -1977,6 +1985,26 @@ static void chipio_8051_read_exram(struct hda_codec *codec, *data = chipio_8051_get_data(codec); } +static void chipio_8051_write_pll_pmu(struct hda_codec *codec, + unsigned int addr, unsigned int data) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + chipio_8051_set_address(codec, addr & 0xff); + chipio_8051_set_data_pll(codec, data); + + mutex_unlock(&spec->chipio_mutex); +} + +static void chipio_8051_write_pll_pmu_no_mutex(struct hda_codec *codec, + unsigned int addr, unsigned int data) +{ + chipio_8051_set_address(codec, addr & 0xff); + chipio_8051_set_data_pll(codec, data); +} + /* * Enable clocks. */ @@ -1985,18 +2013,11 @@ static void chipio_enable_clocks(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; mutex_lock(&spec->chipio_mutex); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 5); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x0b); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 6); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); + + chipio_8051_write_pll_pmu_no_mutex(codec, 0x00, 0xff); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x05, 0x0b); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x06, 0xff); + mutex_unlock(&spec->chipio_mutex); } @@ -7904,10 +7925,7 @@ static void ae5_post_dsp_register_set(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); writeb(0xff, spec->mem_base + 0x304); writeb(0xff, spec->mem_base + 0x304); @@ -7949,30 +7967,11 @@ static void ae5_post_dsp_param_setup(struct hda_codec *codec) static void ae5_post_dsp_pll_setup(struct hda_codec *codec) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x45); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcc); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x40); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcb); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x51); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x8d); + chipio_8051_write_pll_pmu(codec, 0x41, 0xc8); + chipio_8051_write_pll_pmu(codec, 0x45, 0xcc); + chipio_8051_write_pll_pmu(codec, 0x40, 0xcb); + chipio_8051_write_pll_pmu(codec, 0x43, 0xc7); + chipio_8051_write_pll_pmu(codec, 0x51, 0x8d); } static void ae5_post_dsp_stream_setup(struct hda_codec *codec) @@ -7994,10 +7993,7 @@ static void ae5_post_dsp_stream_setup(struct hda_codec *codec) chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 0xc7); ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80); @@ -8090,12 +8086,8 @@ static void ae7_post_dsp_pll_setup(struct hda_codec *codec) }; unsigned int i; - for (i = 0; i < ARRAY_SIZE(addr); i++) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, addr[i]); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, data[i]); - } + for (i = 0; i < ARRAY_SIZE(addr); i++) + chipio_8051_write_pll_pmu_no_mutex(codec, addr[i], data[i]); } static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) @@ -8111,10 +8103,7 @@ static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) mutex_lock(&spec->chipio_mutex); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 0xc7); chipio_write_no_mutex(codec, 0x189000, 0x0001f101); chipio_write_no_mutex(codec, 0x189004, 0x0001f101); @@ -8187,10 +8176,7 @@ static void ae7_post_dsp_asi_setup(struct hda_codec *codec) { chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f); @@ -8207,10 +8193,7 @@ static void ae7_post_dsp_asi_setup(struct hda_codec *codec) ae7_post_dsp_pll_setup(codec); ae7_post_dsp_asi_stream_setup(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu(codec, 0x43, 0xc7); ae7_post_dsp_asi_setup_ports(codec); } @@ -9367,18 +9350,11 @@ static void ae5_register_set(struct hda_codec *codec) unsigned int i, cur_addr; unsigned char tmp[3]; - if (ca0132_quirk(spec) == QUIRK_AE7) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); - } + if (ca0132_quirk(spec) == QUIRK_AE7) + chipio_8051_write_pll_pmu(codec, 0x41, 0xc8); chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); if (ca0132_quirk(spec) == QUIRK_AE7) { tmp[0] = 0x03; @@ -9454,10 +9430,7 @@ static void ca0132_alt_init(struct hda_codec *codec) break; case QUIRK_AE5: ca0132_gpio_init(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + chipio_8051_write_pll_pmu(codec, 0x49, 0x88); chipio_write(codec, 0x18b030, 0x00000020); snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); @@ -9465,10 +9438,7 @@ static void ca0132_alt_init(struct hda_codec *codec) break; case QUIRK_AE7: ca0132_gpio_init(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + chipio_8051_write_pll_pmu(codec, 0x49, 0x88); snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); chipio_write(codec, 0x18b008, 0x000000f8); -- 2.25.1

2 2

[PATCH alsa-utils] alsactl: Remove asound.state file check from alsa-restore.service again
by Takashi Iwai 11 Dec '20

11 Dec '20

We added the check of asound.state file presence some time ago to assure that alsactl gets called only if the state file is already present. Since then, the situation has changed significantly: e.g. now alsactl does initialize if the state file isn't present, and the same alsa-restore.service is used to save the state. This means that we should start this service no matter the state file exists at the boot time or not. So, revert the old change again. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/alsa-restore.service.in | 1 - 1 file changed, 1 deletion(-) diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index a84c2e842444..80fd5fd48203 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -7,7 +7,6 @@ Description=Save/Restore Sound Card State ConditionPathExists=!@daemonswitch@ ConditionPathExistsGlob=/dev/snd/control* -ConditionPathExists=@asoundrcfile@ [Service] Type=oneshot -- 2.26.2

1 0

[PATCH alsa-utils 1/2] alsactl: Fix double decrease of lock timeout
by Takashi Iwai 11 Dec '20

11 Dec '20

The state_lock() has a loop to wait for the lock file creation, and the timeout value gets decremented twice mistakenly, which leads to a half timeout (5 seconds) than expected 10 seconds. Fix it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/lock.c | 1 - 1 file changed, 1 deletion(-) diff --git a/alsactl/lock.c b/alsactl/lock.c index 4a485392b3bd..05f6e4d2a102 100644 --- a/alsactl/lock.c +++ b/alsactl/lock.c @@ -63,7 +63,6 @@ static int state_lock_(const char *file, int lock, int timeout, int _fd) if (fd < 0) { if (errno == EBUSY || errno == EAGAIN) { sleep(1); - timeout--; } else { err = -errno; goto out; -- 2.26.2

1 1

[PATCH 1/2] dt-bindings: add simple-audio-mux binding
by Alexandre Belloni 11 Dec '20

11 Dec '20

Add devicetree documentation for simple audio multiplexers Signed-off-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> --- Cc: Rob Herring <robh+dt(a)kernel.org> .../bindings/sound/simple-audio-mux.yaml | 41 +++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/simple-audio-mux.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml new file mode 100644 index 000000000000..5986d1fcbb54 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-mux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Multiplexer + +maintainers: + - Alexandre Belloni <aleandre.belloni(a)bootlin.com> + +description: | + Simple audio multiplexers are driven using gpios, allowing to select which of + their input line is connected to the output line. + +properties: + compatible: + const: simple-audio-mux + + mux-gpios: + description: | + GPIOs used to select the input line. + + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: + Used as prefix for sink/source names of the component. Must be a + unique string among multiple instances of the same component. + +required: + - compatible + - mux-gpios + +additionalProperties: false + +examples: + - | + mux { + compatible = "simple-audio-mux"; + mux-gpios = <&gpio 3 0>; + }; -- 2.28.0

3 4

[PATCH] alsactl: Skip restore during the lock
by Takashi Iwai 11 Dec '20

11 Dec '20

Currently alsactl-restore tries to initialize the device when an error is found for restore action. But this isn't the right behavior in the case where the lock is held; it implies that another alsactl is running concurrently, hence you shouldn't initialize the card at the same time. The situation is found easily when two alsactls get started by both udev and systemd (note that those two invocations are the designed behavior, see /usr/lib/udev/rules.d/78-sound-cards.rules for details). This patch changes load_state() not to handle the initialization if the locking fails. BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1179904 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/state.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/alsactl/state.c b/alsactl/state.c index ea1d3bcaaddf..8d1d814341bd 100644 --- a/alsactl/state.c +++ b/alsactl/state.c @@ -1663,7 +1663,12 @@ int load_state(const char *file, const char *initfile, int initflags, err = snd_input_stdio_attach(&in, stdin, 0); } else { lock_fd = state_lock(file, 10); - err = lock_fd >= 0 ? snd_input_stdio_open(&in, file, "r") : lock_fd; + /* skip restore if already locked; i.e. concurrent accesses */ + if (lock_fd < 0) { + err = 0; + goto out_global; + } + err = snd_input_stdio_open(&in, file, "r"); } if (err >= 0) { err = snd_config_load(config, in); @@ -1781,6 +1786,7 @@ single: err = finalerr; out: snd_config_delete(config); +out_global: snd_config_update_free_global(); return err; } -- 2.26.2

2 10

[RESEND PATCH 0/2] Convert Tegra HDA doc to json-schema
by Sameer Pujar 11 Dec '20

11 Dec '20

Convert Tegra HDA doc to YAML format. Ran binding validation checks to ensure there are no warnings/errors for HDA. Patch "arm64: tegra: Fix Tegra194 HDA {clock,reset}-names ordering" is already merged and hence not included here. Sameer Pujar (2): dt-bindings: tegra: Convert HDA doc to json-schema dt-bindings: tegra: Add missing HDA properties .../bindings/sound/nvidia,tegra30-hda.txt | 35 ------- .../bindings/sound/nvidia,tegra30-hda.yaml | 112 +++++++++++++++++++++ 2 files changed, 112 insertions(+), 35 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml -- 2.7.4

3 5

[PATCH] ASoC: wm_adsp: remove "ctl" from list on error in wm_adsp_create_control()
by Dan Carpenter 11 Dec '20

11 Dec '20

The error handling frees "ctl" but it's still on the "dsp->ctl_list" list so that could result in a use after free. Remove it from the list before returning. Fixes: 2323736dca72 ("ASoC: wm_adsp: Add basic support for rev 1 firmware file format") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e61d00486c65..dec8716aa8ef 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1519,7 +1519,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); if (!ctl_work) { ret = -ENOMEM; - goto err_ctl_cache; + goto err_list_del; } ctl_work->dsp = dsp; @@ -1529,7 +1529,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, return 0; -err_ctl_cache: +err_list_del: + list_del(&ctl->list); kfree(ctl->cache); err_ctl_subname: kfree(ctl->subname); -- 2.29.2

3 2

[PATCH 1/2] ASoC: topology: Fix wrong size check
by Amadeusz Sławiński 11 Dec '20

11 Dec '20

Dan reported that smatch reports wrong size check and after analysis it is confirmed that we are comparing wrong value: pointer size instead of array size. However the check itself is problematic as in UAPI header there are two fields: struct snd_soc_tplg_enum_control { (...) char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; the texts field is for names and the values one for values assigned to those named fields, after analysis it becomes clear that there is quite a lot overhead values than we may possibly name. So instead of changing check to ARRAY_SIZE(ec->values), as it was first suggested, use hardcoded value of SND_SOC_TPLG_NUM_TEXTS. Link: https://lore.kernel.org/alsa-devel/X9B0eDcKy+9B6kZl@mwanda/ Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index eb2633dd6454..7fb3a87ab860 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -889,10 +889,16 @@ static int soc_tplg_denum_create_values(struct soc_tplg *tplg, struct soc_enum * { int i; - if (le32_to_cpu(ec->items) > sizeof(*ec->values)) + /* + * Following "if" checks if we have at most SND_SOC_TPLG_NUM_TEXTS + * values instead of using ARRAY_SIZE(ec->values) due to the fact that + * it is oversized for its purpose. Additionally it is done so because + * it is defined in UAPI header where it can't be easily changed. + */ + if (le32_to_cpu(ec->items) > SND_SOC_TPLG_NUM_TEXTS) return -EINVAL; - se->dobj.control.dvalues = devm_kzalloc(tplg->dev, le32_to_cpu(ec->items) * + se->dobj.control.dvalues = devm_kcalloc(tplg->dev, le32_to_cpu(ec->items), sizeof(u32), GFP_KERNEL); if (!se->dobj.control.dvalues) -- 2.25.1

3 3

[PATCH v3 0/2] ASoC: rt1015p: delay 300ms for waiting calibration
by Tzung-Bi Shih 11 Dec '20

11 Dec '20

The 1st patch moves SDB control from DAI ops trigger to DAPM event (per review comments in v1). The 2nd patch adds the 300ms delay for waiting calibration. Changes from v2: - Use gpiod_set_value_cansleep() instead of gpiod_set_value(). (https://patchwork.kernel.org/project/alsa-devel/patch/20201210033617.79300-…) - Assuming the calibration state gets lost after system suspend. (https://patchwork.kernel.org/project/alsa-devel/patch/20201210033617.79300-…) Changes from v1: (https://patchwork.kernel.org/project/alsa-devel/patch/20201209033742.382597…) - Move the delay from trigger to DAPM event. Tzung-Bi Shih (2): ASoC: rt1015p: move SDB control from trigger to DAPM ASoC: rt1015p: delay 300ms after SDB pulling high for calibration sound/soc/codecs/rt1015p.c | 69 ++++++++++++++++++-------------------- 1 file changed, 32 insertions(+), 37 deletions(-) -- 2.29.2.684.gfbc64c5ab5-goog

2 3

[PATCH] ASoC: mediatek: mt8183: add PM ops to machine drivers
by Tzung-Bi Shih 11 Dec '20

11 Dec '20

Adds PM ops to machine drivers so that they notify components in the sound card when system suspend. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 + sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 26e7d9a7198f..078e58f1ad0b 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -811,6 +811,7 @@ static struct platform_driver mt8183_da7219_max98357_driver = { #ifdef CONFIG_OF .of_match_table = mt8183_da7219_max98357_dt_match, #endif + .pm = &snd_soc_pm_ops, }, .probe = mt8183_da7219_max98357_dev_probe, }; diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 327dfad41e31..8c8340854859 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -744,6 +744,7 @@ static struct platform_driver mt8183_mt6358_ts3a227_max98357_driver = { #ifdef CONFIG_OF .of_match_table = mt8183_mt6358_ts3a227_max98357_dt_match, #endif + .pm = &snd_soc_pm_ops, }, .probe = mt8183_mt6358_ts3a227_max98357_dev_probe, }; -- 2.29.2.684.gfbc64c5ab5-goog

2 1

[PATCH] ASoC: mediatek: mt8183: delete some unreachable code
by Dan Carpenter 11 Dec '20

11 Dec '20

This has a goto followed by an unreachable return statement. The goto is correct because it cleans up so the current runtime behavior is fine. Let's delete the unreachable return statement. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/mediatek/mt8183/mt8183-afe-clk.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-clk.c b/sound/soc/mediatek/mt8183/mt8183-afe-clk.c index 48e81c5d52fc..cc4f8f4d3dab 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-clk.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-clk.c @@ -584,7 +584,6 @@ int mt8183_mck_enable(struct mtk_base_afe *afe, int mck_id, int rate) __func__, aud_clks[div_clk_id], rate, ret); goto ERR_SET_MCLK_RATE; - return ret; } return 0; -- 2.29.2

2 1

[PATCH 0/3] ASoC: SOF: Intel: fix to dsp state dump trace levels
by Kai Vehmanen 11 Dec '20

11 Dec '20

Small series that addresses a problem where DSP status dump for a failure case, ends up being printed as as debug print. This is important information for any bug report. While at it, the series contains a few cleanups to related code. Ranjani Sridharan (3): ASoC: SOF: Intel: hda: remove duplicated status dump ASoC: SOF: modify the SOF_DBG flags ASoC: SOF: Intel: hda: fix the condition passed to sof_dev_dbg_or_err sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/intel/hda-loader.c | 19 +++++++++++++------ sound/soc/sof/intel/hda.c | 10 ++++------ sound/soc/sof/loader.c | 4 ++-- sound/soc/sof/ops.c | 2 +- sound/soc/sof/sof-priv.h | 13 ++++++++----- 7 files changed, 30 insertions(+), 22 deletions(-) -- 2.29.2

2 4

[PATCH] ASoC: SOF: imx: update kernel-doc description
by Kai Vehmanen 11 Dec '20

11 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add missing parameters to avoid W=1 error Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/imx/imx-common.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index 5fee637834c2..8826ef94f04a 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -47,6 +47,8 @@ void imx8_get_registers(struct snd_sof_dev *sdev, /** * imx8_dump() - This function is called when a panic message is * received from the firmware. + * @sdev: SOF device + * @flags: parameter not used but required by ops prototype */ void imx8_dump(struct snd_sof_dev *sdev, u32 flags) { -- 2.29.2

2 1

[PATCH] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Kai Vehmanen 11 Dec '20

11 Dec '20

The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2ddc27db8c01..81fd7f9a40ce 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -638,11 +638,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.29.2

2 2

[PATCH] ALSA: hda/hdmi: always print pin NIDs as hexadecimal
by Kai Vehmanen 11 Dec '20

11 Dec '20

The debug prints from patch_hdmi.c are not aligned with HDA common code in hda_codec.c nor with other HDA codec drivers. To align with rest of the codebase, use hexadecimal formatting whenever printing value of a HDA NID. Also refer to NIDs with capital letters in traces as is done other modules. This presentation is also aligned with the formatting used in HDA codec procfs entry. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 27 ++++++++++++--------------- 1 file changed, 12 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2ddc27db8c01..22f5a29d25f6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -253,7 +253,7 @@ static int pin_id_to_pin_index(struct hda_codec *codec, return pin_idx; } - codec_warn(codec, "HDMI: pin nid %d not registered\n", pin_nid); + codec_warn(codec, "HDMI: pin NID 0x%x not registered\n", pin_nid); return -EINVAL; } @@ -313,7 +313,7 @@ static int cvt_nid_to_cvt_index(struct hda_codec *codec, hda_nid_t cvt_nid) if (get_cvt(spec, cvt_idx)->cvt_nid == cvt_nid) return cvt_idx; - codec_warn(codec, "HDMI: cvt nid %d not registered\n", cvt_nid); + codec_warn(codec, "HDMI: cvt NID 0x%x not registered\n", cvt_nid); return -EINVAL; } @@ -687,8 +687,7 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, dp_ai->CC02_CT47 = active_channels - 1; dp_ai->CA = ca; } else { - codec_dbg(codec, "HDMI: unknown connection type at pin %d\n", - pin_nid); + codec_dbg(codec, "HDMI: unknown connection type at pin NID 0x%x\n", pin_nid); return; } @@ -701,10 +700,8 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, */ if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, sizeof(ai))) { - codec_dbg(codec, - "hdmi_pin_setup_infoframe: pin=%d channels=%d ca=0x%02x\n", - pin_nid, - active_channels, ca); + codec_dbg(codec, "%s: pin NID=0x%x channels=%d ca=0x%02x\n", + __func__, pin_nid, active_channels, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -796,7 +793,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res, jack->jack_dirty = 1; codec_dbg(codec, - "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + "HDMI hot plug event: Codec=%d NID=0x%x Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, jack->nid, jack->dev_id, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); @@ -874,7 +871,7 @@ static void haswell_verify_D0(struct hda_codec *codec, msleep(40); pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; - codec_dbg(codec, "Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); + codec_dbg(codec, "Haswell HDMI audio: Power for NID 0x%x is now D%d\n", nid, pwr); } } @@ -1121,8 +1118,8 @@ static void intel_not_share_assigned_cvt(struct hda_codec *codec, per_cvt = get_cvt(spec, cvt_idx); if (!per_cvt->assigned) { codec_dbg(codec, - "choose cvt %d for pin nid %d\n", - cvt_idx, nid); + "choose cvt %d for pin NID 0x%x\n", + cvt_idx, nid); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, cvt_idx); @@ -1320,7 +1317,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { codec_warn(codec, - "HDMI: pin %d wcaps %#x does not support connection list\n", + "HDMI: pin NID 0x%x wcaps %#x does not support connection list\n", pin_nid, get_wcaps(codec, pin_nid)); return -EINVAL; } @@ -1635,7 +1632,7 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, eld->eld_valid = false; codec_dbg(codec, - "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", + "HDMI status: Codec=%d NID=0x%x Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { @@ -2795,7 +2792,7 @@ static int intel_pin2port(void *audio_ptr, int pin_nid) return i; } - codec_info(codec, "Can't find the HDMI/DP port for pin %d\n", pin_nid); + codec_info(codec, "Can't find the HDMI/DP port for pin NID 0x%x\n", pin_nid); return -1; } -- 2.29.2

2 1

[PATCH] ALSA: usb-audio: Fix control 'access overflow' errors from chmap
by Takashi Iwai 11 Dec '20

11 Dec '20

The current channel-map control implementation in USB-audio driver may lead to an error message like "control 3:0:0:Playback Channel Map:0: access overflow" when CONFIG_SND_CTL_VALIDATION is set. It's because the chmap get callback clears the whole array no matter which count is set, and rather the false-positive detection. This patch fixes the problem by clearing only the needed array range at usb_chmap_ctl_get(). Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ca76ba5b5c0b..2f6d39c2ba7c 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -193,16 +193,16 @@ static int usb_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct snd_usb_substream *subs = info->private_data; struct snd_pcm_chmap_elem *chmap = NULL; - int i; + int i = 0; - memset(ucontrol->value.integer.value, 0, - sizeof(ucontrol->value.integer.value)); if (subs->cur_audiofmt) chmap = subs->cur_audiofmt->chmap; if (chmap) { for (i = 0; i < chmap->channels; i++) ucontrol->value.integer.value[i] = chmap->map[i]; } + for (; i < subs->channels_max; i++) + ucontrol->value.integer.value[i] = 0; return 0; } -- 2.26.2

1 0

How to annotate USB jacks without jack detection?
by Kai-Heng Feng 11 Dec '20

11 Dec '20

Hi, There are some UAC1 devices come with jacks, but without jack detection ability. And many USB/Thunderbolt docks are equipped with such USB device. Userspace like PulseAudio automatically switch to USB audio when they gets plugged. However, auto-switching to UAC1 device can be problematic because it may not connected to any headset or speaker. So I wonder what's the best way to let userspace know the USB device is a jack without jack detection ability? Through sysfs or dedicated UCM conf? Kai-Heng

2 2

[PATCH] ASoC: fsl: imx-hdmi: Fix an uninitialized return in probe
by Dan Carpenter 11 Dec '20

11 Dec '20

The "ret" variable should be set to a negative error code but currently it returns uninitialized stack data. Fixes: 6a5f850aa83a ("ASoC: fsl: Add imx-hdmi machine driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 2c2a76a71940..ede4a9ad1054 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { dev_err(&pdev->dev, "Invalid HDMI DAI link\n"); + ret = -EINVAL; goto fail; } -- 2.29.2

1 0

Add supported for more Lenovo ALC285 Headset Button -- again
by Kailang 11 Dec '20

11 Dec '20

Hi Takashi, This platform had more SSID. So, use pin quirk for this platform. Many Thanks. Kailang

2 1

Lose HDMI audio and internal DMIC after upgrade from 1.2.3 to 1.2.4 on Intel TigerLake machines
by GitHub issues - opened 11 Dec '20

11 Dec '20

alsa-project/alsa-ucm-conf issue #66 was opened from starnight: We have some laptops equipped with Intel i7-1165G7 and i5-1135G7. The HDMI audio and internal Intel SST DMIC can be detected by PulseAudio with alsa-ucm-conf at **v1.2.3**. But, they are lost when upgrade alsa-ucm-conf to **v1.2.4** (including latest). I did bisect and found it starts to be failed at commit dcef48f ("HDA-Intel: add support for AMD acp microphone devices"). [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/5677934/… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/66 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add supported for more Lenovo ALC285 Headset Button
by Kailang 11 Dec '20

11 Dec '20

Hi Takashi, This platform had more SSID. So, use pin quirk for this platform. Many Thanks. Kailang

1 0

HP Envy 32" snd_hda_codec_realtek
by Mr. Indro Fedrigo 11 Dec '20

11 Dec '20

Hi All, I’ve been trying for a while to get the Audio working on my HP Envy 32 All-in-one desktop. It has a Bang&Olufsen audio system that works perfectly with windows, but it absolutely mute with Ubuntu 20.04- Centos Stream - Fedora 33 - OpenSUSE Tumbleweed... I can get sound output, using Bluetooth, on other devices but nothing from my speakers. I’m attaching the output of alsa-info.sh to begin with, if anybody has any idea. I can provide any other kind of system info if that could help. Thanks. Indro Fedrigo UC Berkeley http://alsa-project.org/db/?f=943238345fb524c3e60f2348b8b39b824d8cdc05

1 0

[PATCH v3 1/2] ALSA: hda/ca0132 - Fix AE-5 rear headphone pincfg.
by Connor McAdams 10 Dec '20

10 Dec '20

The Windows driver sets the pincfg for the AE-5's rear-headphone to report as a microphone. This causes issues with Pulseaudio mistakenly believing there is no headphone plugged in. In Linux, we should instead set it to be a headphone. Fixes: a6b0961b39896 ("ALSA: hda/ca0132 - fix AE-5 pincfg") Cc: <stable(a)kernel.org> Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> Link: https://lore.kernel.org/r/20201208195223.424753-1-conmanx360@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4fbec4258f58..e96db73c32f5 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1223,7 +1223,7 @@ static const struct hda_pintbl ae5_pincfgs[] = { { 0x0e, 0x01c510f0 }, /* SPDIF In */ { 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */ { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ - { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ + { 0x11, 0x012170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ { 0x13, 0x908700f0 }, /* What U Hear In*/ { 0x18, 0x50d000f0 }, /* N/A */ -- 2.25.1

2 2

[PATCH v2 0/2] ASoC: rt1015p: delay 300ms for waiting calibration
by Tzung-Bi Shih 10 Dec '20

10 Dec '20

The 1st patch moves SDB control from DAI ops trigger to DAPM event (per review comments in v1). The 2nd patch adds the 300ms delay for waiting calibration. Changes from v1: (https://patchwork.kernel.org/project/alsa-devel/patch/20201209033742.382597…) - Move the delay from trigger to DAPM event. Tzung-Bi Shih (2): ASoC: rt1015p: move SDB control from trigger to DAPM ASoC: rt1015p: delay 300ms after SDB pulling high for calibration sound/soc/codecs/rt1015p.c | 54 +++++++++++++------------------------- 1 file changed, 18 insertions(+), 36 deletions(-) -- 2.29.2.576.ga3fc446d84-goog

2 8

[PATCH v2 1/3] ALSA: hda/ca0132 - Fix AE-5 rear headphone pincfg.
by Connor McAdams 10 Dec '20

10 Dec '20

The Windows driver sets the pincfg for the AE-5's rear-headphone to report as a microphone. This causes issues with Pulseaudio mistakenly believing there is no headphone plugged in. In Linux, we should instead set it to be a headphone. Fixes: a6b0961b39896 ("ALSA: hda/ca0132 - fix AE-5 pincfg") Cc: <stable(a)kernel.org> Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) v2 changes: - Add fixes references to previous commits that these patches now fix. diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4fbec4258f58..e96db73c32f5 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1223,7 +1223,7 @@ static const struct hda_pintbl ae5_pincfgs[] = { { 0x0e, 0x01c510f0 }, /* SPDIF In */ { 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */ { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ - { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ + { 0x11, 0x012170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ { 0x13, 0x908700f0 }, /* What U Hear In*/ { 0x18, 0x50d000f0 }, /* N/A */ -- 2.25.1

2 7

[PATCH] ALSA: hda/hdmi: fix silent stream for first playback to DP
by Kai Vehmanen 10 Dec '20

10 Dec '20

A problem exists in enabling silent stream when connection type is DisplayPort. Silent stream programming is completed when a new DP receiver is connected, but infoframe transmission does not actually start until PCM is opened for the first time. This can result in audible gap of multiple seconds. This only affects the first PCM open. Fix the issue by properly assigning a converter to the silent stream, and modifying the required stream ID programming sequence. This change only affects Intel display audio codecs. BugLink: https://github.com/thesofproject/linux/issues/2468 Fixes: 951894cf30f4 ("ALSA: hda/hdmi: Add Intel silent stream support") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 94 +++++++++++++++++++++++++++++++++----- 1 file changed, 82 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b0068f8ca46d..6aad2bd09950 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -78,6 +78,7 @@ struct hdmi_spec_per_pin { int pcm_idx; /* which pcm is attached. -1 means no pcm is attached */ int repoll_count; bool setup; /* the stream has been set up by prepare callback */ + bool silent_stream; int channels; /* current number of channels */ bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ @@ -979,6 +980,13 @@ static int hdmi_choose_cvt(struct hda_codec *codec, else per_pin = get_pin(spec, pin_idx); + if (per_pin && per_pin->silent_stream) { + cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); + if (cvt_id) + *cvt_id = cvt_idx; + return 0; + } + /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { per_cvt = get_cvt(spec, cvt_idx); @@ -1642,30 +1650,91 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, snd_hda_power_down_pm(codec); } +#define I915_SILENT_RATE 48000 +#define I915_SILENT_CHANNELS 2 +#define I915_SILENT_FORMAT SNDRV_PCM_FORMAT_S16_LE +#define I915_SILENT_FORMAT_BITS 16 +#define I915_SILENT_FMT_MASK 0xf + static void silent_stream_enable(struct hda_codec *codec, - struct hdmi_spec_per_pin *per_pin) + struct hdmi_spec_per_pin *per_pin) { - unsigned int newval, oldval; - - codec_dbg(codec, "hdmi: enabling silent stream for NID %d\n", - per_pin->pin_nid); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_cvt *per_cvt; + int cvt_idx, pin_idx, err; + unsigned int format; mutex_lock(&per_pin->lock); - if (!per_pin->channels) - per_pin->channels = 2; + if (per_pin->setup) { + codec_dbg(codec, "hdmi: PCM already open, no silent stream\n"); + goto unlock_out; + } + + pin_idx = pin_id_to_pin_index(codec, per_pin->pin_nid, per_pin->dev_id); + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); + if (err) { + codec_err(codec, "hdmi: no free converter to enable silent mode\n"); + goto unlock_out; + } + + per_cvt = get_cvt(spec, cvt_idx); + per_cvt->assigned = 1; + per_pin->cvt_nid = per_cvt->cvt_nid; + per_pin->silent_stream = true; + + codec_dbg(codec, "hdmi: enabling silent stream pin-NID=0x%x cvt-NID=0x%x\n", + per_pin->pin_nid, per_cvt->cvt_nid); - oldval = snd_hda_codec_read(codec, per_pin->pin_nid, 0, - AC_VERB_GET_CONV, 0); - newval = (oldval & 0xF0) | 0xF; - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, newval); + snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + per_pin->mux_idx); + /* configure unused pins to choose other converters */ + pin_cvt_fixup(codec, per_pin, 0); + + snd_hdac_sync_audio_rate(&codec->core, per_pin->pin_nid, + per_pin->dev_id, I915_SILENT_RATE); + + /* trigger silent stream generation in hw */ + format = snd_hdac_calc_stream_format(I915_SILENT_RATE, I915_SILENT_CHANNELS, + I915_SILENT_FORMAT, I915_SILENT_FORMAT_BITS, 0); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, + I915_SILENT_FMT_MASK, I915_SILENT_FMT_MASK, format); + usleep_range(100, 200); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, I915_SILENT_FMT_MASK, 0, format); + + per_pin->channels = I915_SILENT_CHANNELS; hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + unlock_out: mutex_unlock(&per_pin->lock); } +static void silent_stream_disable(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_cvt *per_cvt; + int cvt_idx; + + if (!per_pin->silent_stream) + return; + + codec_dbg(codec, "HDMI: disable silent stream on pin-NID=0x%x cvt-NID=0x%x\n", + per_pin->pin_nid, per_pin->cvt_nid); + + cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); + if (cvt_idx >= 0 && cvt_idx < spec->num_cvts) { + per_cvt = get_cvt(spec, cvt_idx); + per_cvt->assigned = 0; + } + + per_pin->cvt_nid = 0; + per_pin->silent_stream = false; +} + /* update ELD and jack state via audio component */ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) @@ -1701,6 +1770,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, pm_ret); silent_stream_enable(codec, per_pin); } else if (monitor_prev && !monitor_next) { + silent_stream_disable(codec, per_pin); pm_ret = snd_hda_power_down_pm(codec); if (pm_ret < 0) codec_err(codec, -- 2.29.2

2 2

[PATCH 1/3] ALSA: hda/ca0132 - Fix AE-5 rear headphone pincfg.
by Connor McAdams 10 Dec '20

10 Dec '20

The Windows driver sets the pincfg for the AE-5's rear-headphone to report as a microphone. This causes issues with Pulseaudio mistakenly believing there is no headphone plugged in. In Linux, we should instead set it to be a headphone. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4fbec4258f58..e96db73c32f5 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1223,7 +1223,7 @@ static const struct hda_pintbl ae5_pincfgs[] = { { 0x0e, 0x01c510f0 }, /* SPDIF In */ { 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */ { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ - { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ + { 0x11, 0x012170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ { 0x13, 0x908700f0 }, /* What U Hear In*/ { 0x18, 0x50d000f0 }, /* N/A */ -- 2.25.1

2 5

[PATCH v2 0/5] ALSA: hda/ca0132 - Fix no-audio issues and add documentation.
by Connor McAdams 10 Dec '20

10 Dec '20

This patch series attempts to fix the no audio at startup issues that have been occurring for some users, using information gained through reverse engineering the ca0132's onboard 8051 processor. The issue of no audio on startup seemed to be caused by two separate problems: not resetting the codec upon initialization, and not making sure the DSP allocates it's DMA channels correctly. Tested and working on all cards, except for the AE-7. The AE-7 shouldn't have any issues, however, I don't have one on hand to test. Connor McAdams (5): ALSA: hda/ca0132 - Reset codec upon initialization. ALSA: hda/ca0132 - Add stream port remapping function. ALSA: hda/ca0132 - Add 8051 exram helper functions. ALSA: hda/ca0132 - Ensure DSP is properly setup post-firmware download. ALSA: hda/ca0132 - Remove now unnecessary DSP setup functions. sound/pci/hda/patch_ca0132.c | 558 ++++++++++++++++++++++------------- 1 file changed, 354 insertions(+), 204 deletions(-) v2 changes: -Remove conditional mutexes from patches 3 and 4. -- 2.25.1

2 6

[RFC] ASoC: max98373: don't access volatile registers in bias level off
by Bard Liao 10 Dec '20

10 Dec '20

We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To avoid the regmap_read error, this patch try to return a fake value when kcontrol _get is called in suspend. However, the question is what is the "correct" behavior when we try to read a volatile register when cache only is set. 1. return an error code to signal codec is not available as what we have now. 2. return a fake value like what this patch do. 3. wake-up the codec and always return a valid value. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/max98373.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..07494e40c296 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,23 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * We can't get the feedback data when codec is off, set + * value = 0 and return 0 to avoid error. + */ + ucontrol->value.integer.value[0] = 0; + return 0; + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +226,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +245,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), -- 2.17.1

5 7

AMD ACP raven ridge: invalid audio mode 2
by Janpieter Sollie 10 Dec '20

10 Dec '20

Forwarded from: https://bugzilla.kernel.org/show_bug.cgi?id=208699 Anybody who knows what this "ACP" actually does? what's its use? Kind regards, Janpieter Sollie

3 6

[PATCH 1/2] ASoC: Intel: common: add ACPI matching tables for Alder Lake
by Kai Vehmanen 09 Dec '20

09 Dec '20

Initial support for ADL w/ RT711 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 2 +- .../intel/common/soc-acpi-intel-adl-match.c | 52 +++++++++++++++++++ 3 files changed, 55 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-adl-match.c diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 5c49e7d78002..59551b1f22f3 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -29,12 +29,14 @@ 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[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_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 64468fe9c513..12a205ccdeeb 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-cnl-match.o soc-acpi-intel-cfl-match.o \ soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ - soc-acpi-intel-jsl-match.o \ + soc-acpi-intel-jsl-match.o soc-acpi-intel-adl-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-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c new file mode 100644 index 000000000000..06b233d63b73 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -0,0 +1,52 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-apci-intel-adl-match.c - tables and support for ADL ACPI enumeration. + * + * Copyright (c) 2020, Intel Corporation. + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000020025D071100, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr adl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr_d = rt711_0_adr, + }, + {} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); + +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { + { + .link_mask = 0x1, /* link0 required */ + .links = adl_rvp, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt711.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_sdw_machines); -- 2.28.0

2 2

[bug report] ASoC: topology: fix endianness issues
by Dan Carpenter 09 Dec '20

09 Dec '20

[ This bug is way older than your patch but you might know the answer. -dan ] Hello Pierre-Louis Bossart, The patch 5aebe7c7f9c2: "ASoC: topology: fix endianness issues" from Apr 4, 2019, leads to the following static checker warning: sound/soc/soc-topology.c:903 soc_tplg_denum_create_values() warn: potential pointer math issue ('se->dobj.control.dvalues' is a 64 bit pointer) sound/soc/soc-topology.c 887 static int soc_tplg_denum_create_values(struct soc_tplg *tplg, struct soc_enum *se, 888 struct snd_soc_tplg_enum_control *ec) 889 { 890 int i; 891 892 if (le32_to_cpu(ec->items) > sizeof(*ec->values)) The warning is from this line where Smatch starts to think that "ec->items" is the number of bytes, but later in the function we treat it was the number of elements. I do think probably this should be if: if (le32_to_cpu(ec->items) > ARRAY_SIZE(ec->values)) return -EINVAL; The ec->values[] is an array of u32: __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; so this code will return -EINVAL for anything larger than 4. Changing it to ARRAY_SIZE() would raise the limit to 176. 893 return -EINVAL; 894 895 se->dobj.control.dvalues = devm_kzalloc(tplg->dev, le32_to_cpu(ec->items) * 896 sizeof(u32), 897 GFP_KERNEL); 898 if (!se->dobj.control.dvalues) 899 return -ENOMEM; 900 901 /* convert from little-endian */ 902 for (i = 0; i < le32_to_cpu(ec->items); i++) { 903 se->dobj.control.dvalues[i] = le32_to_cpu(ec->values[i]); 904 } 905 906 return 0; 907 } regards, dan carpenter

4 4

[PATCH alsa-utils] alsa-info: Add lsusb and stream outputs
by Takashi Iwai 09 Dec '20

09 Dec '20

We need more detailed information for USB-audio devices, at least the lsusb -v output and the contents of stream* proc files. Let's add them to alsa-info.sh output. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsa-info/alsa-info.sh | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/alsa-info/alsa-info.sh b/alsa-info/alsa-info.sh index f179bfab8655..3871b97a2268 100755 --- a/alsa-info/alsa-info.sh +++ b/alsa-info/alsa-info.sh @@ -476,6 +476,18 @@ cat /proc/asound/card*/codec\#* > $TEMPDIR/alsa-hda-intel.tmp 2> /dev/null cat /proc/asound/card*/codec97\#0/ac97\#0-0 > $TEMPDIR/alsa-ac97.tmp 2> /dev/null cat /proc/asound/card*/codec97\#0/ac97\#0-0+regs > $TEMPDIR/alsa-ac97-regs.tmp 2> /dev/null +#Check for USB descriptors +if [ -x /usr/bin/lsusb ]; then + for f in /proc/asound/card[0-9]*/usbbus; do + test -f "$f" || continue + id=$(sed 's@/@:@' $f) + lsusb -v -s $id >> $TEMPDIR/lsusb.tmp 2> /dev/null + done +fi + +#Check for USB stream setup +cat /proc/asound/card*/stream[0-9]* > $TEMPDIR/alsa-usbstream.tmp 2> /dev/null + #Check for USB mixer setup cat /proc/asound/card*/usbmixer > $TEMPDIR/alsa-usbmixer.tmp 2> /dev/null @@ -649,6 +661,27 @@ if [ -s "$TEMPDIR/alsa-ac97.tmp" ]; then echo "" >> $FILE fi +if [ -s "$TEMPDIR/lsusb.tmp" ]; then + echo "!!USB Descriptors" >> $FILE + echo "!!---------------" >> $FILE + echo "--startcollapse--" >> $FILE + cat $TEMPDIR/lsusb.tmp >> $FILE + echo "--endcollapse--" >> $FILE + echo "" >> $FILE + echo "" >> $FILE +fi + +if [ -s "$TEMPDIR/lsusb.tmp" ]; then + echo "!!USB Stream information" >> $FILE + echo "!!----------------------" >> $FILE + echo "--startcollapse--" >> $FILE + echo "" >> $FILE + cat $TEMPDIR/alsa-usbstream.tmp >> $FILE + echo "--endcollapse--" >> $FILE + echo "" >> $FILE + echo "" >> $FILE +fi + if [ -s "$TEMPDIR/alsa-usbmixer.tmp" ]; then echo "!!USB Mixer information" >> $FILE echo "!!---------------------" >> $FILE -- 2.26.2

1 0

[PATCH] ALSA: usb-audio: Add implicit fb support for Steinberg UR22
by Takashi Iwai 09 Dec '20

09 Dec '20

Steinberg UR22 (with USB ID 0499:1509) requires the implicit feedback for the proper playback, otherwise it causes occasional cracks. This patch adds the corresponding the quirk table entry with the recently added generic implicit fb support. Reported-and-tested-by: Kilian <meschi(a)posteo.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 386198b36b87..4e911d200562 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -42,6 +42,7 @@ struct snd_usb_implicit_fb_match { /* Implicit feedback quirk table for playback */ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Generic matching */ + IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2080), /* M-Audio FastTrack Ultra */ IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2081), /* M-Audio FastTrack Ultra */ IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ -- 2.26.2

1 0

Steinberg UR22 Crackling and Clicks
by meschi＠posteo.de 09 Dec '20

09 Dec '20

Hi, I am using a Steinberg UR22 audio interface. Since installing Ubuntu 20.04 on my workstation I have slight crackling in audio playback. It doesn't matter if the audio gets routed through pulse, jack or played straight to alsa. When running jack, the crackling is not showing any xruns. It's not possible to get any glitch-free playback in the system. The onboard intel audio works fine. My system: alsa info: alsa-project.org/db/?f=8b821b17a8c83fa5db866f34a5b4dfe74d640a32 Processor: Intel Xeon X5687 $ uname -a Linux WorkStation-T5500 5.4.0-56-lowlatency #62-Ubuntu SMP PREEMPT Mon Nov 23 20:19:44 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux I have tried: 5.4.0-56-generic and -lowlatency 5.8.0-31-generic and -lowlatency setting cpu governor to performance using nouveau driver over nvidia When playing a file with aplay like this: systemctl --user stop pulseaudio.socket systemctl --user stop pulseaudio.service aplay -D plughw:CARD=UR22 ./test.wav slight crackling can be heard every 5-10 seconds. It also appears, that the clicks heard are not always on both channels, sometimes the left stereo channel clicks, sometimes the right, sometimes both. Plugging the Interface into different USB-ports doesn't help. The only external devices connected via USB are my mouse and my keyboard. On Wndows the same setup on the same machine works fine. To make things even weirder, using the same audio interface on my Thinkpad x220 (Core i5-2540M) with ubuntu 20.04 and 5.4.0.56-lowlatency does not show this problem. Running dmesg seems to show nothing audio driver related after: [ 22.425520] usbcore: registered new interface driver snd-usb-audio I have found a similar issue showcasing the problem on manjaro with a differend soundcard. Therefore I suspect the problem is caused by ALSA interacting with the hardware. https://forum.manjaro.org/t/pops-clicks-on-audio-interface/40129 Here is another occurence on ubuntu on yet another audio interface. https://askubuntu.com/questions/1256787/ubuntu-2004-novation-xiosynth-exter… If I can supply more information I'm happy to help. I couldn't find a similar issue in this mailing list, so if this is already fixed I apologize in advance. Kilian

3 15

[PATCH 1/5] ASoC: codecs/jz47xx: Use regmap_{set,clear}_bits
by Paul Cercueil 09 Dec '20

09 Dec '20

Use regmap_{set,clear}_bits instead of regmap_update_bits, when applicable. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- sound/soc/codecs/jz4725b.c | 26 ++++++------ sound/soc/codecs/jz4740.c | 20 +++------ sound/soc/codecs/jz4770.c | 85 +++++++++++++++++--------------------- 3 files changed, 55 insertions(+), 76 deletions(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index e49374c72e70..5201a8f6d7b6 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -198,15 +198,15 @@ static int jz4725b_out_stage_enable(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - return regmap_update_bits(map, JZ4725B_CODEC_REG_IFR, - BIT(REG_IFR_RAMP_UP_DONE_OFFSET), 0); + return regmap_clear_bits(map, JZ4725B_CODEC_REG_IFR, + BIT(REG_IFR_RAMP_UP_DONE_OFFSET)); case SND_SOC_DAPM_POST_PMU: return regmap_read_poll_timeout(map, JZ4725B_CODEC_REG_IFR, val, val & BIT(REG_IFR_RAMP_UP_DONE_OFFSET), 100000, 500000); case SND_SOC_DAPM_PRE_PMD: - return regmap_update_bits(map, JZ4725B_CODEC_REG_IFR, - BIT(REG_IFR_RAMP_DOWN_DONE_OFFSET), 0); + return regmap_clear_bits(map, JZ4725B_CODEC_REG_IFR, + BIT(REG_IFR_RAMP_DOWN_DONE_OFFSET)); case SND_SOC_DAPM_POST_PMD: return regmap_read_poll_timeout(map, JZ4725B_CODEC_REG_IFR, val, val & BIT(REG_IFR_RAMP_DOWN_DONE_OFFSET), @@ -303,24 +303,22 @@ static int jz4725b_codec_set_bias_level(struct snd_soc_component *component, switch (level) { case SND_SOC_BIAS_ON: - regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, - BIT(REG_PMR2_SB_SLEEP_OFFSET), 0); + regmap_clear_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_SLEEP_OFFSET)); break; case SND_SOC_BIAS_PREPARE: /* Enable sound hardware */ - regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, - BIT(REG_PMR2_SB_OFFSET), 0); + regmap_clear_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_OFFSET)); msleep(224); break; case SND_SOC_BIAS_STANDBY: - regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, - BIT(REG_PMR2_SB_SLEEP_OFFSET), - BIT(REG_PMR2_SB_SLEEP_OFFSET)); + regmap_set_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_SLEEP_OFFSET)); break; case SND_SOC_BIAS_OFF: - regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, - BIT(REG_PMR2_SB_OFFSET), - BIT(REG_PMR2_SB_OFFSET)); + regmap_set_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_OFFSET)); break; } diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index c9900d1cd5c2..5e58bfee2b49 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -219,12 +219,11 @@ static struct snd_soc_dai_driver jz4740_codec_dai = { static void jz4740_codec_wakeup(struct regmap *regmap) { - regmap_update_bits(regmap, JZ4740_REG_CODEC_1, - JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET); + regmap_set_bits(regmap, JZ4740_REG_CODEC_1, JZ4740_CODEC_1_RESET); udelay(2); - regmap_update_bits(regmap, JZ4740_REG_CODEC_1, - JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0); + regmap_clear_bits(regmap, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET); regcache_sync(regmap); } @@ -235,7 +234,6 @@ static int jz4740_codec_set_bias_level(struct snd_soc_component *component, struct jz4740_codec *jz4740_codec = snd_soc_component_get_drvdata(component); struct regmap *regmap = jz4740_codec->regmap; unsigned int mask; - unsigned int value; switch (level) { case SND_SOC_BIAS_ON: @@ -244,9 +242,8 @@ static int jz4740_codec_set_bias_level(struct snd_soc_component *component, mask = JZ4740_CODEC_1_VREF_DISABLE | JZ4740_CODEC_1_VREF_AMP_DISABLE | JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; - value = 0; - regmap_update_bits(regmap, JZ4740_REG_CODEC_1, mask, value); + regmap_clear_bits(regmap, JZ4740_REG_CODEC_1, mask); break; case SND_SOC_BIAS_STANDBY: /* The only way to clear the suspend flag is to reset the codec */ @@ -256,17 +253,12 @@ static int jz4740_codec_set_bias_level(struct snd_soc_component *component, mask = JZ4740_CODEC_1_VREF_DISABLE | JZ4740_CODEC_1_VREF_AMP_DISABLE | JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; - value = JZ4740_CODEC_1_VREF_DISABLE | - JZ4740_CODEC_1_VREF_AMP_DISABLE | - JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; - regmap_update_bits(regmap, JZ4740_REG_CODEC_1, mask, value); + regmap_set_bits(regmap, JZ4740_REG_CODEC_1, mask); break; case SND_SOC_BIAS_OFF: mask = JZ4740_CODEC_1_SUSPEND; - value = JZ4740_CODEC_1_SUSPEND; - - regmap_update_bits(regmap, JZ4740_REG_CODEC_1, mask, value); + regmap_set_bits(regmap, JZ4740_REG_CODEC_1, mask); regcache_mark_dirty(regmap); break; default: diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index 298689a07168..c6b2043f31a9 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -190,18 +190,18 @@ static int jz4770_codec_set_bias_level(struct snd_soc_component *codec, switch (level) { case SND_SOC_BIAS_PREPARE: - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC, - REG_CR_VIC_SB, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_CR_VIC, + REG_CR_VIC_SB); msleep(250); - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC, - REG_CR_VIC_SB_SLEEP, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_CR_VIC, + REG_CR_VIC_SB_SLEEP); msleep(400); break; case SND_SOC_BIAS_STANDBY: - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC, - REG_CR_VIC_SB_SLEEP, REG_CR_VIC_SB_SLEEP); - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC, - REG_CR_VIC_SB, REG_CR_VIC_SB); + regmap_set_bits(regmap, JZ4770_CODEC_REG_CR_VIC, + REG_CR_VIC_SB_SLEEP); + regmap_set_bits(regmap, JZ4770_CODEC_REG_CR_VIC, + REG_CR_VIC_SB); fallthrough; default: break; @@ -292,8 +292,8 @@ static int jz4770_codec_mute_stream(struct snd_soc_dai *dai, int mute, int direc } /* clear GUP/GDO flag */ - regmap_update_bits(jz_codec->regmap, JZ4770_CODEC_REG_IFR, - gain_bit, gain_bit); + regmap_set_bits(jz_codec->regmap, JZ4770_CODEC_REG_IFR, + gain_bit); } return 0; @@ -369,8 +369,8 @@ static int hpout_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: /* set cap-less, unmute HP */ - regmap_update_bits(jz_codec->regmap, JZ4770_CODEC_REG_CR_HP, - REG_CR_HP_SB_HPCM | REG_CR_HP_MUTE, 0); + regmap_clear_bits(jz_codec->regmap, JZ4770_CODEC_REG_CR_HP, + REG_CR_HP_SB_HPCM | REG_CR_HP_MUTE); break; case SND_SOC_DAPM_POST_PMU: @@ -385,16 +385,15 @@ static int hpout_event(struct snd_soc_dapm_widget *w, } /* clear RUP flag */ - regmap_update_bits(jz_codec->regmap, JZ4770_CODEC_REG_IFR, - REG_IFR_RUP, REG_IFR_RUP); + regmap_set_bits(jz_codec->regmap, JZ4770_CODEC_REG_IFR, + REG_IFR_RUP); break; case SND_SOC_DAPM_POST_PMD: /* set cap-couple, mute HP */ - regmap_update_bits(jz_codec->regmap, JZ4770_CODEC_REG_CR_HP, - REG_CR_HP_SB_HPCM | REG_CR_HP_MUTE, - REG_CR_HP_SB_HPCM | REG_CR_HP_MUTE); + regmap_set_bits(jz_codec->regmap, JZ4770_CODEC_REG_CR_HP, + REG_CR_HP_SB_HPCM | REG_CR_HP_MUTE); err = regmap_read_poll_timeout(jz_codec->regmap, JZ4770_CODEC_REG_IFR, @@ -406,8 +405,8 @@ static int hpout_event(struct snd_soc_dapm_widget *w, } /* clear RDO flag */ - regmap_update_bits(jz_codec->regmap, JZ4770_CODEC_REG_IFR, - REG_IFR_RDO, REG_IFR_RDO); + regmap_set_bits(jz_codec->regmap, JZ4770_CODEC_REG_IFR, + REG_IFR_RDO); break; } @@ -592,63 +591,53 @@ static void jz4770_codec_codec_init_regs(struct snd_soc_component *codec) regcache_cache_only(regmap, true); /* default HP output to PCM */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_HP, - REG_CR_HP_SEL_MASK, REG_CR_HP_SEL_MASK); + regmap_set_bits(regmap, JZ4770_CODEC_REG_CR_HP, REG_CR_HP_SEL_MASK); /* default line output to PCM */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_LO, - REG_CR_LO_SEL_MASK, REG_CR_LO_SEL_MASK); + regmap_set_bits(regmap, JZ4770_CODEC_REG_CR_LO, REG_CR_LO_SEL_MASK); /* Disable stereo mic */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_MIC, - BIT(REG_CR_MIC_STEREO_OFFSET), 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_CR_MIC, + BIT(REG_CR_MIC_STEREO_OFFSET)); /* Set mic 1 as default source for ADC */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_ADC, - REG_CR_ADC_IN_SEL_MASK, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_CR_ADC, + REG_CR_ADC_IN_SEL_MASK); /* ADC/DAC: serial + i2s */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_AICR_ADC, - REG_AICR_ADC_SERIAL | REG_AICR_ADC_I2S, - REG_AICR_ADC_SERIAL | REG_AICR_ADC_I2S); - regmap_update_bits(regmap, JZ4770_CODEC_REG_AICR_DAC, - REG_AICR_DAC_SERIAL | REG_AICR_DAC_I2S, - REG_AICR_DAC_SERIAL | REG_AICR_DAC_I2S); + regmap_set_bits(regmap, JZ4770_CODEC_REG_AICR_ADC, + REG_AICR_ADC_SERIAL | REG_AICR_ADC_I2S); + regmap_set_bits(regmap, JZ4770_CODEC_REG_AICR_DAC, + REG_AICR_DAC_SERIAL | REG_AICR_DAC_I2S); /* The generated IRQ is a high level */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_ICR, - REG_ICR_INT_FORM_MASK, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_ICR, REG_ICR_INT_FORM_MASK); regmap_update_bits(regmap, JZ4770_CODEC_REG_IMR, REG_IMR_ALL_MASK, REG_IMR_JACK_MASK | REG_IMR_RUP_MASK | REG_IMR_RDO_MASK | REG_IMR_GUP_MASK | REG_IMR_GDO_MASK); /* 12M oscillator */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CCR, - REG_CCR_CRYSTAL_MASK, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_CCR, REG_CCR_CRYSTAL_MASK); /* 0: 16ohm/220uF, 1: 10kohm/1uF */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_HP, - REG_CR_HP_LOAD, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_CR_HP, REG_CR_HP_LOAD); /* disable automatic gain */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_AGC1, REG_AGC1_EN, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_AGC1, REG_AGC1_EN); /* Disable DAC lrswap */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_DAC, - REG_CR_DAC_LRSWAP, REG_CR_DAC_LRSWAP); + regmap_set_bits(regmap, JZ4770_CODEC_REG_CR_DAC, REG_CR_DAC_LRSWAP); /* Independent L/R DAC gain control */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_GCR_DACL, - REG_GCR_DACL_RLGOD, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_GCR_DACL, + REG_GCR_DACL_RLGOD); /* Disable ADC lrswap */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_ADC, - REG_CR_ADC_LRSWAP, REG_CR_ADC_LRSWAP); + regmap_set_bits(regmap, JZ4770_CODEC_REG_CR_ADC, REG_CR_ADC_LRSWAP); /* default to cap-less mode(0) */ - regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_HP, - REG_CR_HP_SB_HPCM, 0); + regmap_clear_bits(regmap, JZ4770_CODEC_REG_CR_HP, REG_CR_HP_SB_HPCM); /* Send collected updates. */ regcache_cache_only(regmap, false); -- 2.29.2

2 5

[PATCH] ASoC: Intel: bytcr_rt5640: Add quirk for ARCHOS Cesium 140
by Chris Chiu 09 Dec '20

09 Dec '20

Tha ARCHOS Cesium 140 tablet has problem with the jack-sensing, thus the heaset functions are not working. Add quirk for this model to select the correct input map, jack-detect options and channel map to enable jack sensing and headset microphone. This device uses IN1 for its internal MIC and JD2 for jack-detect. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f790514a147d..cd6f7caa43c8 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -421,6 +421,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.20.1

3 2

[PATCH v2] ASoC: AMD Raven/Renoir - fix the PCI probe (PCI revision)
by Jaroslav Kysela 09 Dec '20

09 Dec '20

The Raven and Renoir ACP can be distinguished by the PCI revision. Let's do the check very early, otherwise the wrong probe code can be run. Link: https://lore.kernel.org/alsa-devel/2e4587f8-f602-cf23-4845-fd27a32b1cfc@amd… Cc: <stable(a)kernel.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/raven/pci-acp3x.c | 4 ++++ sound/soc/amd/renoir/rn-pci-acp3x.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 31b797c8bfe6..77f2d9389606 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -118,6 +118,10 @@ static int snd_acp3x_probe(struct pci_dev *pci, int ret, i; u32 addr, val; + /* Raven device detection */ + if (pci->revision != 0x00) + return -ENODEV; + if (pci_enable_device(pci)) { dev_err(&pci->dev, "pci_enable_device failed\n"); return -ENODEV; diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 60a3e2f442db..338b78c514ec 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -188,6 +188,10 @@ static int snd_rn_acp_probe(struct pci_dev *pci, int ret, index; u32 addr; + /* Renoir device check */ + if (pci->revision != 0x01) + return -ENODEV; + if (pci_enable_device(pci)) { dev_err(&pci->dev, "pci_enable_device failed\n"); return -ENODEV; -- 2.26.2

2 1

[PATCH] ASoC: max98390: Fix error codes in max98390_dsm_init()
by Dan Carpenter 09 Dec '20

09 Dec '20

These error paths return success but they should return -EINVAL. Fixes: 97ed3e509ee6 ("ASoC: max98390: Fix potential crash during param fw loading") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/max98390.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index ff5cc9bbec29..bb736c44e68a 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -784,6 +784,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) { dev_err(component->dev, "param fw is invalid.\n"); + ret = -EINVAL; goto err_alloc; } dsm_param = (char *)fw->data; @@ -794,6 +795,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { dev_err(component->dev, "param fw is invalid.\n"); + ret = -EINVAL; goto err_alloc; } regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); -- 2.29.2

2 1

[PATCH] ASoC: rt1015: check the return value of regmap_read during i2c probe
by derek.fang＠realtek.com 09 Dec '20

09 Dec '20

From: Derek Fang <derek.fang(a)realtek.com> In some projects, the device ID register is not read correctly. This patch helps to verify the issue is caused from i2c host or client. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index ac4c9f43b338..32e6bcf763d1 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1207,8 +1207,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, rt1015->hw_config = (i2c->addr == 0x29) ? RT1015_HW_29 : RT1015_HW_28; - regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); - if ((val != RT1015_DEVICE_ID_VAL) && (val != RT1015_DEVICE_ID_VAL2)) { + ret = regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); + if (ret) { + dev_err(&i2c->dev, + "Failed to read device register: %d\n", ret); + return ret; + } else if ((val != RT1015_DEVICE_ID_VAL) && + (val != RT1015_DEVICE_ID_VAL2)) { dev_err(&i2c->dev, "Device with ID register %x is not rt1015\n", val); return -ENODEV; -- 2.17.1

2 1

[PATCH] ALSA: hda: Fix regressions on clear and reconfig sysfs
by Takashi Iwai 09 Dec '20

09 Dec '20

It seems that the HD-audio clear and reconfig sysfs don't work any longer after the recent driver core change. There are multiple issues around that: the linked list corruption and the dead device handling. The former issue is fixed by another patch for the driver core itself, while the latter patch needs to be addressed in HD-audio side. This patch corresponds to the latter, it recovers those broken functions by replacing the device detach and attach actions with the standard core API functions, which are almost equivalent with unbind and bind actions. Fixes: 654888327e9f ("driver core: Avoid binding drivers to dead devices") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=209207 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4bb58e8b08a8..687216e74526 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1803,7 +1803,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) return -EBUSY; /* OK, let it free */ - snd_hdac_device_unregister(&codec->core); + device_release_driver(hda_codec_dev(codec)); /* allow device access again */ snd_hda_unlock_devices(bus); diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index eb8ec109d7ad..d5ffcba794e5 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -139,7 +139,7 @@ static int reconfig_codec(struct hda_codec *codec) "The codec is being used, can't reconfigure.\n"); goto error; } - err = snd_hda_codec_configure(codec); + err = device_reprobe(hda_codec_dev(codec)); if (err < 0) goto error; err = snd_card_register(codec->card); -- 2.26.2

1 0

Re: Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Takashi Iwai 09 Dec '20

09 Dec '20

On Wed, 09 Dec 2020 13:16:23 +0100, Ben Bell wrote: > > [...] > > > and learning abobut it led me to conclude that the Wing needs an entry in > > > the quirks list in set_sync_ep_implicit_fb_quirk to properly enable it: > > > > > > --- sound/usb/pcm.c.orig 2020-11-22 23:36:08.000000000 +0000 > > > +++ sound/usb/pcm.c 2020-12-05 08:40:21.639600074 +0000 > > > @@ -340,6 +345,7 @@ > > > ep = 0x81; > > > ifnum = 3; > > > goto add_sync_ep_from_ifnum; > > > + case USB_ID(0x1397, 0x050b): /* Behringer Wing */ > > > case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ > > > case USB_ID(0x0763, 0x2081): > > > > > > A week's worth of debugging and learning yielded a one line patch ;) > > > > That's a interesting piece of information :) > > > > Could you give lsusb -v output of the device? With the latest patch > > set, the driver can set up the implicit feedback mode automatically in > > certain cases. I'm not sure whether Behringer device follows that > > pattern. If not, we'd need to add the quirk manually. > > I did wonder whether it was possible to automate this, as all I did > was looked out the endpoint with the Implicit Feedback flag set. > Fingers crossed it'll work. > > Output attached. Thanks. Through a quick glance, this would work as is without patching. The implicit fb source EP is found in the next interface with the same altset as the playback sink EP. Let's cross fingers :) Takashi > > bjb > > > > Bus 009 Device 005: ID 1397:050b BEHRINGER International GmbH > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > idVendor 0x1397 BEHRINGER International GmbH > idProduct 0x050b > bcdDevice 1.02 > iManufacturer 1 BEHRINGER > iProduct 3 WING > iSerial 2 WING-PP-09110504 > bNumConfigurations 2 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 0x0323 > bNumInterfaces 5 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0xc0 > Self Powered > MaxPower 0mA > Interface Association: > bLength 8 > bDescriptorType 11 > bFirstInterface 0 > bInterfaceCount 3 > bFunctionClass 1 Audio > bFunctionSubClass 0 > bFunctionProtocol 32 > iFunction 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 1 Control Device > bInterfaceProtocol 32 > iInterface 3 WING > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 2.00 > bCategory 8 > wTotalLength 0x01df > bmControls 0x00 > AudioControl Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 10 (CLOCK_SOURCE) > bClockID 41 > bmAttributes 1 Internal fixed clock > bmControls 0x07 > Clock Frequency Control (read/write) > Clock Validity Control (read-only) > bAssocTerminal 0 > iClockSource 9 WING Internal Clock > AudioControl Interface Descriptor: > bLength 17 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 2 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bCSourceID 41 > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 18 Out 1 > bmControls 0x0000 > iTerminal 6 WING Output > AudioControl Interface Descriptor: > bLength 202 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 10 > bSourceID 2 > bmaControls(0) 0x00000000 > bmaControls(1) 0x00000000 > bmaControls(2) 0x00000000 > bmaControls(3) 0x00000000 > bmaControls(4) 0x00000000 > bmaControls(5) 0x00000000 > bmaControls(6) 0x00000000 > bmaControls(7) 0x00000000 > bmaControls(8) 0x00000000 > bmaControls(9) 0x00000000 > bmaControls(10) 0x00000000 > bmaControls(11) 0x00000000 > bmaControls(12) 0x00000000 > bmaControls(13) 0x00000000 > bmaControls(14) 0x00000000 > bmaControls(15) 0x00000000 > bmaControls(16) 0x00000000 > bmaControls(17) 0x00000000 > bmaControls(18) 0x00000000 > bmaControls(19) 0x00000000 > bmaControls(20) 0x00000000 > bmaControls(21) 0x00000000 > bmaControls(22) 0x00000000 > bmaControls(23) 0x00000000 > bmaControls(24) 0x00000000 > bmaControls(25) 0x00000000 > bmaControls(26) 0x00000000 > bmaControls(27) 0x00000000 > bmaControls(28) 0x00000000 > bmaControls(29) 0x00000000 > bmaControls(30) 0x00000000 > bmaControls(31) 0x00000000 > bmaControls(32) 0x00000000 > bmaControls(33) 0x00000000 > bmaControls(34) 0x00000000 > bmaControls(35) 0x00000000 > bmaControls(36) 0x00000000 > bmaControls(37) 0x00000000 > bmaControls(38) 0x00000000 > bmaControls(39) 0x00000000 > bmaControls(40) 0x00000000 > bmaControls(41) 0x00000000 > bmaControls(42) 0x00000000 > bmaControls(43) 0x00000000 > bmaControls(44) 0x00000000 > bmaControls(45) 0x00000000 > bmaControls(46) 0x00000000 > bmaControls(47) 0x00000000 > bmaControls(48) 0x00000000 > iFeature 0 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 20 > wTerminalType 0x0301 Speaker > bAssocTerminal 0 > bSourceID 10 > bCSourceID 41 > bmControls 0x0000 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 17 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 1 > wTerminalType 0x0201 Microphone > bAssocTerminal 0 > bCSourceID 41 > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 66 In 1 > bmControls 0x0000 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 202 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 11 > bSourceID 1 > bmaControls(0) 0x00000000 > bmaControls(1) 0x00000000 > bmaControls(2) 0x00000000 > bmaControls(3) 0x00000000 > bmaControls(4) 0x00000000 > bmaControls(5) 0x00000000 > bmaControls(6) 0x00000000 > bmaControls(7) 0x00000000 > bmaControls(8) 0x00000000 > bmaControls(9) 0x00000000 > bmaControls(10) 0x00000000 > bmaControls(11) 0x00000000 > bmaControls(12) 0x00000000 > bmaControls(13) 0x00000000 > bmaControls(14) 0x00000000 > bmaControls(15) 0x00000000 > bmaControls(16) 0x00000000 > bmaControls(17) 0x00000000 > bmaControls(18) 0x00000000 > bmaControls(19) 0x00000000 > bmaControls(20) 0x00000000 > bmaControls(21) 0x00000000 > bmaControls(22) 0x00000000 > bmaControls(23) 0x00000000 > bmaControls(24) 0x00000000 > bmaControls(25) 0x00000000 > bmaControls(26) 0x00000000 > bmaControls(27) 0x00000000 > bmaControls(28) 0x00000000 > bmaControls(29) 0x00000000 > bmaControls(30) 0x00000000 > bmaControls(31) 0x00000000 > bmaControls(32) 0x00000000 > bmaControls(33) 0x00000000 > bmaControls(34) 0x00000000 > bmaControls(35) 0x00000000 > bmaControls(36) 0x00000000 > bmaControls(37) 0x00000000 > bmaControls(38) 0x00000000 > bmaControls(39) 0x00000000 > bmaControls(40) 0x00000000 > bmaControls(41) 0x00000000 > bmaControls(42) 0x00000000 > bmaControls(43) 0x00000000 > bmaControls(44) 0x00000000 > bmaControls(45) 0x00000000 > bmaControls(46) 0x00000000 > bmaControls(47) 0x00000000 > bmaControls(48) 0x00000000 > iFeature 0 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 22 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bSourceID 11 > bCSourceID 41 > bmControls 0x0000 > iTerminal 7 WING Input > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 4 WING Output Terminal > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 4 WING Output Terminal > AudioStreaming Interface Descriptor: > bLength 16 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 2 > bmControls 0x00 > bFormatType 1 > bmFormats 0x00000001 > PCM > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 18 Out 1 > AudioStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bSubslotSize 3 > bBitResolution 24 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x03f0 1x 1008 bytes > bInterval 1 > AudioStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x00 > bmControls 0x00 > bLockDelayUnits 2 Decoded PCM samples > wLockDelay 0x0008 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 5 WING Input Terminal > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 5 WING Input Terminal > AudioStreaming Interface Descriptor: > bLength 16 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 22 > bmControls 0x00 > bFormatType 1 > bmFormats 0x00000001 > PCM > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 66 In 1 > AudioStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bSubslotSize 3 > bBitResolution 24 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x81 EP 1 IN > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x03f0 1x 1008 bytes > bInterval 1 > AudioStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x00 > bmControls 0x00 > bLockDelayUnits 2 Decoded PCM samples > wLockDelay 0x0008 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 1 Control Device > bInterfaceProtocol 0 > iInterface 0 > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 0x0009 > bInCollection 1 > baInterfaceNr(0) 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 4 > bAlternateSetting 0 > bNumEndpoints 2 > bInterfaceClass 1 Audio > bInterfaceSubClass 3 MIDI Streaming > bInterfaceProtocol 0 > iInterface 0 > MIDIStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 0x00a1 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 25 > iJack 11 WING MIDI DAW 1 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 2 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 26 > iJack 13 WING MIDI DAW 2 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 3 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 27 > iJack 15 WING MIDI DAW 3 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 4 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 28 > iJack 17 WING MIDI Control > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 9 > bNrInputPins 1 > baSourceID( 0) 25 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 17 > bNrInputPins 1 > baSourceID( 0) 1 > BaSourcePin( 0) 1 > iJack 10 WING MIDI DAW 1 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 10 > bNrInputPins 1 > baSourceID( 0) 26 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 18 > bNrInputPins 1 > baSourceID( 0) 2 > BaSourcePin( 0) 1 > iJack 12 WING MIDI DAW 2 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 11 > bNrInputPins 1 > baSourceID( 0) 27 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 19 > bNrInputPins 1 > baSourceID( 0) 3 > BaSourcePin( 0) 1 > iJack 14 WING MIDI DAW 3 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 12 > bNrInputPins 1 > baSourceID( 0) 28 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 20 > bNrInputPins 1 > baSourceID( 0) 4 > BaSourcePin( 0) 1 > iJack 16 WING MIDI Control > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x02 EP 2 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > bRefresh 0 > bSynchAddress 0 > MIDIStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 4 > baAssocJackID( 0) 1 > baAssocJackID( 1) 2 > baAssocJackID( 2) 3 > baAssocJackID( 3) 4 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > bRefresh 0 > bSynchAddress 0 > MIDIStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 4 > baAssocJackID( 0) 9 > baAssocJackID( 1) 10 > baAssocJackID( 2) 11 > baAssocJackID( 3) 12 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 0x0323 > bNumInterfaces 5 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0xc0 > Self Powered > MaxPower 0mA > Interface Association: > bLength 8 > bDescriptorType 11 > bFirstInterface 0 > bInterfaceCount 3 > bFunctionClass 1 Audio > bFunctionSubClass 0 > bFunctionProtocol 32 > iFunction 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 1 Control Device > bInterfaceProtocol 32 > iInterface 3 WING > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 2.00 > bCategory 8 > wTotalLength 0x01df > bmControls 0x00 > AudioControl Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 10 (CLOCK_SOURCE) > bClockID 41 > bmAttributes 1 Internal fixed clock > bmControls 0x07 > Clock Frequency Control (read/write) > Clock Validity Control (read-only) > bAssocTerminal 0 > iClockSource 9 WING Internal Clock > AudioControl Interface Descriptor: > bLength 17 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 2 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bCSourceID 41 > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 18 Out 1 > bmControls 0x0000 > iTerminal 6 WING Output > AudioControl Interface Descriptor: > bLength 202 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 10 > bSourceID 2 > bmaControls(0) 0x00000000 > bmaControls(1) 0x00000000 > bmaControls(2) 0x00000000 > bmaControls(3) 0x00000000 > bmaControls(4) 0x00000000 > bmaControls(5) 0x00000000 > bmaControls(6) 0x00000000 > bmaControls(7) 0x00000000 > bmaControls(8) 0x00000000 > bmaControls(9) 0x00000000 > bmaControls(10) 0x00000000 > bmaControls(11) 0x00000000 > bmaControls(12) 0x00000000 > bmaControls(13) 0x00000000 > bmaControls(14) 0x00000000 > bmaControls(15) 0x00000000 > bmaControls(16) 0x00000000 > bmaControls(17) 0x00000000 > bmaControls(18) 0x00000000 > bmaControls(19) 0x00000000 > bmaControls(20) 0x00000000 > bmaControls(21) 0x00000000 > bmaControls(22) 0x00000000 > bmaControls(23) 0x00000000 > bmaControls(24) 0x00000000 > bmaControls(25) 0x00000000 > bmaControls(26) 0x00000000 > bmaControls(27) 0x00000000 > bmaControls(28) 0x00000000 > bmaControls(29) 0x00000000 > bmaControls(30) 0x00000000 > bmaControls(31) 0x00000000 > bmaControls(32) 0x00000000 > bmaControls(33) 0x00000000 > bmaControls(34) 0x00000000 > bmaControls(35) 0x00000000 > bmaControls(36) 0x00000000 > bmaControls(37) 0x00000000 > bmaControls(38) 0x00000000 > bmaControls(39) 0x00000000 > bmaControls(40) 0x00000000 > bmaControls(41) 0x00000000 > bmaControls(42) 0x00000000 > bmaControls(43) 0x00000000 > bmaControls(44) 0x00000000 > bmaControls(45) 0x00000000 > bmaControls(46) 0x00000000 > bmaControls(47) 0x00000000 > bmaControls(48) 0x00000000 > iFeature 0 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 20 > wTerminalType 0x0301 Speaker > bAssocTerminal 0 > bSourceID 10 > bCSourceID 41 > bmControls 0x0000 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 17 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 1 > wTerminalType 0x0201 Microphone > bAssocTerminal 0 > bCSourceID 41 > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 66 In 1 > bmControls 0x0000 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 202 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 11 > bSourceID 1 > bmaControls(0) 0x00000000 > bmaControls(1) 0x00000000 > bmaControls(2) 0x00000000 > bmaControls(3) 0x00000000 > bmaControls(4) 0x00000000 > bmaControls(5) 0x00000000 > bmaControls(6) 0x00000000 > bmaControls(7) 0x00000000 > bmaControls(8) 0x00000000 > bmaControls(9) 0x00000000 > bmaControls(10) 0x00000000 > bmaControls(11) 0x00000000 > bmaControls(12) 0x00000000 > bmaControls(13) 0x00000000 > bmaControls(14) 0x00000000 > bmaControls(15) 0x00000000 > bmaControls(16) 0x00000000 > bmaControls(17) 0x00000000 > bmaControls(18) 0x00000000 > bmaControls(19) 0x00000000 > bmaControls(20) 0x00000000 > bmaControls(21) 0x00000000 > bmaControls(22) 0x00000000 > bmaControls(23) 0x00000000 > bmaControls(24) 0x00000000 > bmaControls(25) 0x00000000 > bmaControls(26) 0x00000000 > bmaControls(27) 0x00000000 > bmaControls(28) 0x00000000 > bmaControls(29) 0x00000000 > bmaControls(30) 0x00000000 > bmaControls(31) 0x00000000 > bmaControls(32) 0x00000000 > bmaControls(33) 0x00000000 > bmaControls(34) 0x00000000 > bmaControls(35) 0x00000000 > bmaControls(36) 0x00000000 > bmaControls(37) 0x00000000 > bmaControls(38) 0x00000000 > bmaControls(39) 0x00000000 > bmaControls(40) 0x00000000 > bmaControls(41) 0x00000000 > bmaControls(42) 0x00000000 > bmaControls(43) 0x00000000 > bmaControls(44) 0x00000000 > bmaControls(45) 0x00000000 > bmaControls(46) 0x00000000 > bmaControls(47) 0x00000000 > bmaControls(48) 0x00000000 > iFeature 0 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 22 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bSourceID 11 > bCSourceID 41 > bmControls 0x0000 > iTerminal 7 WING Input > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 4 WING Output Terminal > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 4 WING Output Terminal > AudioStreaming Interface Descriptor: > bLength 16 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 2 > bmControls 0x00 > bFormatType 1 > bmFormats 0x00000001 > PCM > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 18 Out 1 > AudioStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bSubslotSize 3 > bBitResolution 24 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x03f0 1x 1008 bytes > bInterval 1 > AudioStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x00 > bmControls 0x00 > bLockDelayUnits 2 Decoded PCM samples > wLockDelay 0x0008 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 5 WING Input Terminal > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 32 > iInterface 5 WING Input Terminal > AudioStreaming Interface Descriptor: > bLength 16 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 22 > bmControls 0x00 > bFormatType 1 > bmFormats 0x00000001 > PCM > bNrChannels 48 > bmChannelConfig 0x00000000 > iChannelNames 66 In 1 > AudioStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bSubslotSize 3 > bBitResolution 24 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x81 EP 1 IN > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x03f0 1x 1008 bytes > bInterval 1 > AudioStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x00 > bmControls 0x00 > bLockDelayUnits 2 Decoded PCM samples > wLockDelay 0x0008 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 1 Control Device > bInterfaceProtocol 0 > iInterface 0 > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 0x0009 > bInCollection 1 > baInterfaceNr(0) 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 4 > bAlternateSetting 0 > bNumEndpoints 2 > bInterfaceClass 1 Audio > bInterfaceSubClass 3 MIDI Streaming > bInterfaceProtocol 0 > iInterface 0 > MIDIStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 0x00a1 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 25 > iJack 11 WING MIDI DAW 1 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 2 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 26 > iJack 13 WING MIDI DAW 2 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 3 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 27 > iJack 15 WING MIDI DAW 3 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 4 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 28 > iJack 17 WING MIDI Control > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 9 > bNrInputPins 1 > baSourceID( 0) 25 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 17 > bNrInputPins 1 > baSourceID( 0) 1 > BaSourcePin( 0) 1 > iJack 10 WING MIDI DAW 1 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 10 > bNrInputPins 1 > baSourceID( 0) 26 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 18 > bNrInputPins 1 > baSourceID( 0) 2 > BaSourcePin( 0) 1 > iJack 12 WING MIDI DAW 2 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 11 > bNrInputPins 1 > baSourceID( 0) 27 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 19 > bNrInputPins 1 > baSourceID( 0) 3 > BaSourcePin( 0) 1 > iJack 14 WING MIDI DAW 3 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 1 Embedded > bJackID 12 > bNrInputPins 1 > baSourceID( 0) 28 > BaSourcePin( 0) 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > bJackType 2 External > bJackID 20 > bNrInputPins 1 > baSourceID( 0) 4 > BaSourcePin( 0) 1 > iJack 16 WING MIDI Control > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x02 EP 2 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > bRefresh 0 > bSynchAddress 0 > MIDIStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 4 > baAssocJackID( 0) 1 > baAssocJackID( 1) 2 > baAssocJackID( 2) 3 > baAssocJackID( 3) 4 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > bRefresh 0 > bSynchAddress 0 > MIDIStreaming Endpoint Descriptor: > bLength 8 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 4 > baAssocJackID( 0) 9 > baAssocJackID( 1) 10 > baAssocJackID( 2) 11 > baAssocJackID( 3) 12 > Device Qualifier (for other device speed): > bLength 10 > bDescriptorType 6 > bcdUSB 2.00 > bDeviceClass 0 > bDeviceSubClass 0 > bDeviceProtocol 0 > bMaxPacketSize0 64 > bNumConfigurations 1 > Device Status: 0x0001 > Self Powered

1 0

Re: Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Takashi Iwai 09 Dec '20

09 Dec '20

On Sat, 05 Dec 2020 15:43:56 +0100, Ben Bell wrote: > > > > Like I said in the initial email, it feels like some sort of clock drift > > > problem, where we're managing very slowly to collect more samples than > > > we're reading -- to the tune of about 1 extra every few cycles -- and > > > nothing on the consumer side is ever managing to compensate for that. > > > I'm not even sure how that sort of drift would be possible though. Seems > > > surprising. > [...] > > > At least you can try the latest patch set destined for 5.11, which > > should improve the cases for the implicit feedback. > > Aha! This actually was the key piece of information I needed. I haven't > tried 5.11 or the latest patch set yet, but googling "implicit feedback" > and learning abobut it led me to conclude that the Wing needs an entry in > the quirks list in set_sync_ep_implicit_fb_quirk to properly enable it: > > --- sound/usb/pcm.c.orig 2020-11-22 23:36:08.000000000 +0000 > +++ sound/usb/pcm.c 2020-12-05 08:40:21.639600074 +0000 > @@ -340,6 +345,7 @@ > ep = 0x81; > ifnum = 3; > goto add_sync_ep_from_ifnum; > + case USB_ID(0x1397, 0x050b): /* Behringer Wing */ > case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ > case USB_ID(0x0763, 0x2081): > > > A week's worth of debugging and learning yielded a one line patch ;) That's a interesting piece of information :) Could you give lsusb -v output of the device? With the latest patch set, the driver can set up the implicit feedback mode automatically in certain cases. I'm not sure whether Behringer device follows that pattern. If not, we'd need to add the quirk manually. thanks, Takashi

2 1

[PATCH] ASoC: rt1015p: delay 300ms after SDB pulling high for calibration
by Tzung-Bi Shih 09 Dec '20

09 Dec '20

RT1015p needs 300ms delay after SDB pulling high for internal calibration during the power on sequence. Delays 300ms right before data sends out to avoid data truncated. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/codecs/rt1015p.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 59bb60682270..515357474eba 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -4,6 +4,7 @@ // // Copyright 2020 The Linux Foundation. All rights reserved. +#include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> #include <linux/gpio.h> @@ -20,6 +21,7 @@ struct rt1015p_priv { struct gpio_desc *sdb; int sdb_switch; + bool calib_done; }; static int rt1015p_daiops_trigger(struct snd_pcm_substream *substream, @@ -39,6 +41,11 @@ static int rt1015p_daiops_trigger(struct snd_pcm_substream *substream, if (rt1015p->sdb_switch) { gpiod_set_value(rt1015p->sdb, 1); dev_dbg(component->dev, "set sdb to 1"); + + if (!rt1015p->calib_done) { + mdelay(300); + rt1015p->calib_done = true; + } } break; case SNDRV_PCM_TRIGGER_STOP: -- 2.29.2.576.ga3fc446d84-goog

3 2

[PATCH] ASoC: rt1015: check the return value of regmap_read during i2c probe
by derek.fang＠realtek.com 09 Dec '20

09 Dec '20

From: "derek.fang" <derek(a)realtek.com> In some projects, the device ID register is not read correctly. This patch helps to verify the issue is caused from i2c host or client. Signed-off-by: derek.fang <derek(a)realtek.com> diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index ac4c9f43b338..32e6bcf763d1 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1207,8 +1207,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, rt1015->hw_config = (i2c->addr == 0x29) ? RT1015_HW_29 : RT1015_HW_28; - regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); - if ((val != RT1015_DEVICE_ID_VAL) && (val != RT1015_DEVICE_ID_VAL2)) { + ret = regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); + if (ret) { + dev_err(&i2c->dev, + "Failed to read device register: %d\n", ret); + return ret; + } else if ((val != RT1015_DEVICE_ID_VAL) && + (val != RT1015_DEVICE_ID_VAL2)) { dev_err(&i2c->dev, "Device with ID register %x is not rt1015\n", val); return -ENODEV; -- 2.17.1

1 0

[PATCH 1/2] ALSA: usb-audio: Fix potential out-of-bounds shift
by Takashi Iwai 09 Dec '20

09 Dec '20

syzbot spotted a potential out-of-bounds shift in the USB-audio format parser that receives the arbitrary shift value from the USB descriptor. Add a range check for avoiding the undefined behavior. Reported-by: syzbot+df7dc146ebdd6435eea3(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 93459ba228d3..9ebc5d202c87 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -39,6 +39,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + if (format >= 64) + return 0; /* invalid format */ sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; format = 1ULL << format; -- 2.26.2

1 1

[PATCH 0/5] ALSA: hda/ca0132 - Fix no-audio issues and add documentation.
by Connor McAdams 09 Dec '20

09 Dec '20

This patch series attempts to fix the no audio at startup issues that have been occurring for some users, using information gained through reverse engineering the ca0132's onboard 8051 processor. The issue of no audio on startup seemed to be caused by two separate problems: not resetting the codec upon initialization, and not making sure the DSP allocates it's DMA channels correctly. Tested and working on all cards, except for the AE-7. The AE-7 shouldn't have any issues, however, I don't have one on hand to test. As an aside, I'm not sure how to classify which commits this necessarily 'fixes', as it's covering multiple areas. Please let me know if this is necessary. Connor McAdams (5): ALSA: hda/ca0132 - Reset codec upon initialization. ALSA: hda/ca0132 - Add stream port remapping function. ALSA: hda/ca0132 - Add 8051 exram helper functions. ALSA: hda/ca0132 - Ensure DSP is properly setup post-firmware download. ALSA: hda/ca0132 - Remove now unnecessary DSP setup functions. sound/pci/hda/patch_ca0132.c | 567 ++++++++++++++++++++++------------- 1 file changed, 363 insertions(+), 204 deletions(-) -- 2.25.1

2 6

[PATCH] ALSA: hda/realtek - Enable headset mic of ASUS Q524UQK with ALC255
by Chris Chiu 09 Dec '20

09 Dec '20

The ASUS laptop Q524UQK with ALC255 codec can't detect the headset microphone until ALC255_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..2d4bb9019cb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7976,6 +7976,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), -- 2.20.1

2 1

UBSAN: shift-out-of-bounds in snd_pcm_oss_change_params_locked
by syzbot 09 Dec '20

09 Dec '20

Hello, syzbot found the following issue on: HEAD commit: 15ac8fdb Add linux-next specific files for 20201207 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=1656cc13500000 kernel config: https://syzkaller.appspot.com/x/.config?x=3696b8138207d24d dashboard link: https://syzkaller.appspot.com/bug?extid=33ef0b6639a8d2d42b4c compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=13a8ad37500000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=15bc6adf500000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+33ef0b6639a8d2d42b4c(a)syzkaller.appspotmail.com ================================================================================ UBSAN: shift-out-of-bounds in sound/core/oss/pcm_oss.c:705:23 shift exponent 58 is too large for 32-bit type 'int' CPU: 1 PID: 8476 Comm: syz-executor572 Not tainted 5.10.0-rc6-next-20201207-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x107/0x163 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:395 snd_pcm_oss_period_size sound/core/oss/pcm_oss.c:705 [inline] snd_pcm_oss_change_params_locked.cold+0x55/0x78 sound/core/oss/pcm_oss.c:925 snd_pcm_oss_change_params sound/core/oss/pcm_oss.c:1084 [inline] snd_pcm_oss_make_ready+0xe7/0x1b0 sound/core/oss/pcm_oss.c:1143 snd_pcm_oss_sync+0x1de/0x800 sound/core/oss/pcm_oss.c:1708 snd_pcm_oss_release+0x276/0x300 sound/core/oss/pcm_oss.c:2546 __fput+0x283/0x920 fs/file_table.c:280 task_work_run+0xdd/0x190 kernel/task_work.c:140 exit_task_work include/linux/task_work.h:30 [inline] do_exit+0xb89/0x2a00 kernel/exit.c:823 do_group_exit+0x125/0x310 kernel/exit.c:920 __do_sys_exit_group kernel/exit.c:931 [inline] __se_sys_exit_group kernel/exit.c:929 [inline] __x64_sys_exit_group+0x3a/0x50 kernel/exit.c:929 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x43ee98 Code: Unable to access opcode bytes at RIP 0x43ee6e. RSP: 002b:00007ffc0b9ddff8 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 000000000043ee98 RDX: 0000000000000000 RSI: 000000000000003c RDI: 0000000000000000 RBP: 00000000004be6a8 R08: 00000000000000e7 R09: ffffffffffffffd0 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000001 R13: 00000000006d0180 R14: 0000000000000000 R15: 0000000000000000 ================================================================================ --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

UBSAN: shift-out-of-bounds in parse_audio_format_i
by syzbot 09 Dec '20

09 Dec '20

Hello, syzbot found the following issue on: HEAD commit: 15ac8fdb Add linux-next specific files for 20201207 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=16d4620f500000 kernel config: https://syzkaller.appspot.com/x/.config?x=3696b8138207d24d dashboard link: https://syzkaller.appspot.com/bug?extid=df7dc146ebdd6435eea3 compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=126afa13500000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=14389613500000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+df7dc146ebdd6435eea3(a)syzkaller.appspotmail.com usb 1-1: New USB device found, idVendor=1d6b, idProduct=0101, bcdDevice= 0.40 usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz usb 1-1: 2:1 : no or invalid class specific endpoint descriptor ================================================================================ UBSAN: shift-out-of-bounds in sound/usb/format.c:44:17 shift exponent 4098 is too large for 64-bit type 'long long unsigned int' CPU: 0 PID: 8656 Comm: kworker/0:4 Not tainted 5.10.0-rc6-next-20201207-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x107/0x163 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:395 parse_audio_format_i_type sound/usb/format.c:44 [inline] parse_audio_format_i.cold+0xba/0x3e2 sound/usb/format.c:653 snd_usb_parse_audio_format+0x89/0x290 sound/usb/format.c:753 snd_usb_get_audioformat_uac12 sound/usb/stream.c:841 [inline] __snd_usb_parse_audio_interface+0xce4/0x3cf0 sound/usb/stream.c:1170 snd_usb_parse_audio_interface+0x79/0x130 sound/usb/stream.c:1240 snd_usb_create_stream.isra.0+0x23a/0x530 sound/usb/card.c:206 snd_usb_create_streams sound/usb/card.c:278 [inline] usb_audio_probe+0x93c/0x2ab0 sound/usb/card.c:802 usb_probe_interface+0x315/0x7f0 drivers/usb/core/driver.c:396 really_probe+0x2b1/0xe40 drivers/base/dd.c:554 driver_probe_device+0x285/0x3f0 drivers/base/dd.c:738 __device_attach_driver+0x216/0x2d0 drivers/base/dd.c:844 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4c0 drivers/base/dd.c:912 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbb2/0x1ce0 drivers/base/core.c:2934 usb_set_configuration+0x113c/0x1910 drivers/usb/core/message.c:2167 usb_generic_driver_probe+0xba/0x100 drivers/usb/core/generic.c:238 usb_probe_device+0xd9/0x2c0 drivers/usb/core/driver.c:293 really_probe+0x2b1/0xe40 drivers/base/dd.c:554 driver_probe_device+0x285/0x3f0 drivers/base/dd.c:738 __device_attach_driver+0x216/0x2d0 drivers/base/dd.c:844 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4c0 drivers/base/dd.c:912 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbb2/0x1ce0 drivers/base/core.c:2934 usb_new_device.cold+0x725/0x1057 drivers/usb/core/hub.c:2555 hub_port_connect drivers/usb/core/hub.c:5223 [inline] hub_port_connect_change drivers/usb/core/hub.c:5363 [inline] port_event drivers/usb/core/hub.c:5509 [inline] hub_event+0x2348/0x42d0 drivers/usb/core/hub.c:5591 process_one_work+0x98d/0x1630 kernel/workqueue.c:2275 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421 kthread+0x3b1/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:296 ================================================================================ --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[PATCH v2] ALSA: hda/proc - print DP-MST connections
by Kai Vehmanen 08 Dec '20

08 Dec '20

To help in debugging issues with DisplayPort Multi-Stream Transport (aka DP-MST) support, print information of active connections for each device of a display audio pin widget. Example output with the patch with two monitors connected to a DP-MST hub: Devices: 4 Dev 00: PD = 0, ELDV = 0, IA = 0, Connections [ 0x03* 0x05 0x07 0x09 ] Dev 01: PD = 1, ELDV = 1, IA = 0, Connections [ 0x03* 0x05 0x07 0x09 ] *Dev 02: PD = 1, ELDV = 1, IA = 0, Connections [ 0x03 0x05* 0x07 0x09 ] Dev 03: PD = 0, ELDV = 0, IA = 0, Connections [ 0x03* 0x05 0x07 0x09 ] Connection: 4 0x03 0x05* 0x07 0x09 Format of existing "Connection:" entry is left intact to keep compatibility. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_proc.c | 38 +++++++++++++++++++++++++++++++++++++- 1 file changed, 37 insertions(+), 1 deletion(-) v2 changes: - add NULL check to conn array allocation - put a few functions to a single line as they fit within 100 columns - reformat the first paragraph of commit message diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0631f31ef87f..00c2eeb2c472 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -679,6 +679,38 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_dpmst_connections(struct snd_info_buffer *buffer, struct hda_codec *codec, + hda_nid_t nid, int dev_num) +{ + int c, conn_len, curr, dev_id_saved; + hda_nid_t *conn; + + conn_len = snd_hda_get_num_raw_conns(codec, nid); + if (conn_len <= 0) + return; + + conn = kmalloc_array(conn_len, sizeof(hda_nid_t), GFP_KERNEL); + if (!conn) + return; + + dev_id_saved = snd_hda_get_dev_select(codec, nid); + + snd_hda_set_dev_select(codec, nid, dev_num); + curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (snd_hda_get_raw_connections(codec, nid, conn, conn_len) < 0) + goto out; + + for (c = 0; c < conn_len; c++) { + snd_iprintf(buffer, " 0x%02x", conn[c]); + if (c == curr) + snd_iprintf(buffer, "*"); + } + +out: + kfree(conn); + snd_hda_set_dev_select(codec, nid, dev_id_saved); +} + static void print_device_list(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { @@ -702,10 +734,14 @@ static void print_device_list(struct snd_info_buffer *buffer, snd_iprintf(buffer, " "); snd_iprintf(buffer, - "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i, + "Dev %02d: PD = %d, ELDV = %d, IA = %d, Connections [", i, !!(dev_list[i] & AC_DE_PD), !!(dev_list[i] & AC_DE_ELDV), !!(dev_list[i] & AC_DE_IA)); + + print_dpmst_connections(buffer, codec, nid, i); + + snd_iprintf(buffer, " ]\n"); } } -- 2.28.0

2 1

[PATCH] ASoC: AMD Renoir - add DMI table to avoid the ACP mic probe (broken BIOS)
by Jaroslav Kysela 08 Dec '20

08 Dec '20

Users reported that some Lenovo AMD platforms do not have ACP microphone, but the BIOS advertises it via ACPI. This patch create a simple DMI table, where those machines with the broken BIOS can be added. The DMI description for Lenovo IdeaPad 5 and IdeaPad Flex 5 devices are added there. Also describe the dmic_acpi_check kernel module parameter in a more understandable way. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 Cc: <stable(a)kernel.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index b943e59fc302..3289ab3eae6f 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -6,6 +6,7 @@ #include <linux/pci.h> #include <linux/acpi.h> +#include <linux/dmi.h> #include <linux/module.h> #include <linux/io.h> #include <linux/delay.h> @@ -20,14 +21,13 @@ module_param(acp_power_gating, int, 0644); MODULE_PARM_DESC(acp_power_gating, "Enable acp power gating"); /** - * dmic_acpi_check = -1 - Checks ACPI method to know DMIC hardware status runtime - * = 0 - Skips the DMIC device creation and returns probe failure - * = 1 - Assumes that platform has DMIC support and skips ACPI - * method check + * dmic_acpi_check = -1 - Use ACPI/DMI method to detect the DMIC hardware presence at runtime + * = 0 - Skip the DMIC device creation and return probe failure + * = 1 - Force DMIC support */ static int dmic_acpi_check = ACP_DMIC_AUTO; module_param(dmic_acpi_check, bint, 0644); -MODULE_PARM_DESC(dmic_acpi_check, "checks Dmic hardware runtime"); +MODULE_PARM_DESC(dmic_acpi_check, "Digital microphone presence (-1=auto, 0=none, 1=force)"); struct acp_dev_data { void __iomem *acp_base; @@ -163,6 +163,17 @@ static int rn_acp_deinit(void __iomem *acp_base) return 0; } +static const struct dmi_system_id rn_acp_quirk_table[] = { + { + /* Lenovo IdeaPad Flex 5 14ARE05, IdeaPad 5 15ARE05 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"), + } + }, + {} +}; + static int snd_rn_acp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -172,6 +183,7 @@ static int snd_rn_acp_probe(struct pci_dev *pci, acpi_handle handle; acpi_integer dmic_status; #endif + const struct dmi_system_id *dmi_id; unsigned int irqflags; int ret, index; u32 addr; @@ -232,6 +244,12 @@ static int snd_rn_acp_probe(struct pci_dev *pci, goto de_init; } #endif + dmi_id = dmi_first_match(rn_acp_quirk_table); + if (dmi_id && !dmi_id->driver_data) { + dev_warn(&pci->dev, "ACPI settings override using DMI (ACP mic is not present)"); + ret = -ENODEV; + goto de_init; + } } adata->res = devm_kzalloc(&pci->dev, -- 2.26.2

4 10

[PATCH] ASoC: AMD Raven/Renoir - fix the PCI probe (PCI revision)
by Jaroslav Kysela 08 Dec '20

08 Dec '20

The Raven and Renoir ACP can be distinguished by the PCI revision. Let's do the check very early, otherwise the wrong probe code can be run. Link: https://lore.kernel.org/alsa-devel/2e4587f8-f602-cf23-4845-fd27a32b1cfc@amd… Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/raven/pci-acp3x.c | 4 ++++ sound/soc/amd/renoir/rn-pci-acp3x.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 31b797c8bfe6..77f2d9389606 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -118,6 +118,10 @@ static int snd_acp3x_probe(struct pci_dev *pci, int ret, i; u32 addr, val; + /* Raven device detection */ + if (pci->revision != 0x00) + return -ENODEV; + if (pci_enable_device(pci)) { dev_err(&pci->dev, "pci_enable_device failed\n"); return -ENODEV; diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 60a3e2f442db..338b78c514ec 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -188,6 +188,10 @@ static int snd_rn_acp_probe(struct pci_dev *pci, int ret, index; u32 addr; + /* Renoir device check */ + if (pci->revision != 0x01) + return -ENODEV; + if (pci_enable_device(pci)) { dev_err(&pci->dev, "pci_enable_device failed\n"); return -ENODEV; -- 2.26.2

1 0

[PATCH v2] ASoC: amd: change clk_get() to devm_clk_get() and add missed checks
by Chuhong Yuan 08 Dec '20

08 Dec '20

cz_da7219_init() does not check the return values of clk_get(), while da7219_clk_enable() calls clk_set_rate() to dereference the pointers. Add checks to fix the problems. Also, change clk_get() to devm_clk_get() to avoid data leak after failures. Fixes: bb24a31ed584 ("ASoC: AMD: Configure wclk and bclk of master codec") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- Changes in v2: - Replace clk_get() with devm_clk_get() to prevent data leak. sound/soc/amd/acp-da7219-max98357a.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a7702e64ec51..849288d01c6b 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -73,8 +73,13 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); - da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = devm_clk_get(component->dev, "da7219-dai-wclk"); + if (IS_ERR(da7219_dai_wclk)) + return PTR_ERR(da7219_dai_wclk); + + da7219_dai_bclk = devm_clk_get(component->dev, "da7219-dai-bclk"); + if (IS_ERR(da7219_dai_bclk)) + return PTR_ERR(da7219_dai_bclk); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | -- 2.26.2

2 1

[PATCH v2 1/2] ASoC: dt-bindings: imx-hdmi: Add binding doc for hdmi machine driver
by Shengjiu Wang 08 Dec '20

08 Dec '20

Imx-hdmi is a new added machine driver for supporting hdmi devices on i.MX platforms. There is HDMI IP or external HDMI modules connect with SAI or AUD2HTX interface. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - none .../bindings/sound/imx-audio-hdmi.yaml | 52 +++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml new file mode 100644 index 000000000000..d5474f83ac2c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-hdmi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio complex with HDMI + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-hdmi + - fsl,imx-audio-sii902x + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-cpu: + description: The phandle of an CPU DAI controller + + hdmi-out: + description: | + This is a boolean property. If present, the transmitting function + of HDMI will be enabled, indicating there's a physical HDMI out + connector or jack on the board or it's connecting to some other IP + block, such as an HDMI encoder or display-controller. + + hdmi-in: + description: | + This is a boolean property. If present, the receiving function of + HDMI will be enabled, indicating there is a physical HDMI in + connector/jack on the board. + +required: + - compatible + - model + - audio-cpu + +additionalProperties: false + +examples: + - | + sound-hdmi { + compatible = "fsl,imx-audio-hdmi"; + model = "audio-hdmi"; + audio-cpu = <&aud2htx>; + hdmi-out; + }; -- 2.27.0

3 3

[PATCH 0/2] ASoC: Intel: cht_bsw_nau8824: 2 fixes for usage with sof-audio-acpi
by Hans de Goede 08 Dec '20

08 Dec '20

Hi All, Here are 2 simple fixes which are necessary to make the cht_bsw_nau8824 machine driver work together with the sof-audio-acpi driver. Note that atm the sof topology files are missing a .tplg file for this setup. Simply copying over the standard sof-byte-codec.tplg file does the trick, but then some mixer setting changes are necessary to fix the right speaker/headphones channel not working; and those mixer settings break the right channel when used with the sst-acpi driver. I've been trying to fix this at the tplg level so that we do not need to change the mixer settings, but no luck sofar. I'll post a RFC with the topology changes which I have and we can discuss this further there. These 2 simple fixes are necessary to make the sof-audio-acpi driver work regardless of the topology issue. Regards, Hans

3 4

[PATCH] ASoC: cx2072x: Fix doubly definitions of Playback and Capture streams
by Takashi Iwai 08 Dec '20

08 Dec '20

The cx2072x codec driver defines multiple DAIs with the same stream name "Playback" and "Capture". Although the current code works more or less as is as the secondary streams are never used, it still leads the error message like: debugfs: File 'Playback' in directory 'dapm' already present! debugfs: File 'Capture' in directory 'dapm' already present! Fix it by renaming the secondary streams to unique names. Fixes: a497a4363706 ("ASoC: Add support for Conexant CX2072X CODEC") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/cx2072x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index 2ad00ed21bec..2f10991a8bdb 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -1579,7 +1579,7 @@ static struct snd_soc_dai_driver soc_codec_cx2072x_dai[] = { .id = CX2072X_DAI_DSP, .probe = cx2072x_dsp_dai_probe, .playback = { - .stream_name = "Playback", + .stream_name = "DSP Playback", .channels_min = 2, .channels_max = 2, .rates = CX2072X_RATES_DSP, @@ -1591,7 +1591,7 @@ static struct snd_soc_dai_driver soc_codec_cx2072x_dai[] = { .name = "cx2072x-aec", .id = 3, .capture = { - .stream_name = "Capture", + .stream_name = "AEC Capture", .channels_min = 2, .channels_max = 2, .rates = CX2072X_RATES_DSP, -- 2.26.2

2 1

[PATCH] ALSA: hda/proc - print DP-MST connections
by Kai Vehmanen 08 Dec '20

08 Dec '20

To help in debugging issues with DisplayPort Multi-Stream Transport (aka DP-MST) support, print information of active connections for each device of a display audio pin widget. Example output with the patch with two monitors connected to a DP-MST hub: Devices: 4 Dev 00: PD = 0, ELDV = 0, IA = 0, Connections [ 0x03* 0x05 0x07 0x09 ] Dev 01: PD = 1, ELDV = 1, IA = 0, Connections [ 0x03* 0x05 0x07 0x09 ] *Dev 02: PD = 1, ELDV = 1, IA = 0, Connections [ 0x03 0x05* 0x07 0x09 ] Dev 03: PD = 0, ELDV = 0, IA = 0, Connections [ 0x03* 0x05 0x07 0x09 ] Connection: 4 0x03 0x05* 0x07 0x09 Format of existing "Connection:" entry is left intact to keep compatibility. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_proc.c | 39 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 38 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0631f31ef87f..e1cba574cadb 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -679,6 +679,39 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_dpmst_connections(struct snd_info_buffer *buffer, struct hda_codec *codec, + hda_nid_t nid, int dev_num) +{ + int c, conn_len, curr, dev_id_saved; + hda_nid_t *conn; + + conn_len = snd_hda_get_num_raw_conns(codec, nid); + if (conn_len <= 0) + return; + + conn = kmalloc_array(conn_len, + sizeof(hda_nid_t), + GFP_KERNEL); + + dev_id_saved = snd_hda_get_dev_select(codec, nid); + snd_hda_set_dev_select(codec, nid, dev_num); + if (snd_hda_get_raw_connections(codec, nid, conn, conn_len) < 0) + goto out; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + + for (c = 0; c < conn_len; c++) { + snd_iprintf(buffer, " 0x%02x", conn[c]); + if (c == curr) + snd_iprintf(buffer, "*"); + } + +out: + kfree(conn); + snd_hda_set_dev_select(codec, nid, dev_id_saved); +} + static void print_device_list(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { @@ -702,10 +735,14 @@ static void print_device_list(struct snd_info_buffer *buffer, snd_iprintf(buffer, " "); snd_iprintf(buffer, - "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i, + "Dev %02d: PD = %d, ELDV = %d, IA = %d, Connections [", i, !!(dev_list[i] & AC_DE_PD), !!(dev_list[i] & AC_DE_ELDV), !!(dev_list[i] & AC_DE_IA)); + + print_dpmst_connections(buffer, codec, nid, i); + + snd_iprintf(buffer, " ]\n"); } } -- 2.28.0

2 1

[PATCH v6 0/6] Tegra210 audio graph card
by Sameer Pujar 08 Dec '20

08 Dec '20

This series adds audio graph based sound card support for Tegra210 platforms like Jetson-TX1 an Jetson-Nano. The following preparatory audio graph enhancement series is already merged. * https://patchwork.kernel.org/project/alsa-devel/list/?series=375629&state=* Following are the summary of changes: * Add graph/audio-graph based schemas or schema updates for Tegra210 component and machine drivers. * Add Tegra audio graph machine driver. * Add required DT support for Jetson-TX1/Nano. This work is based on earlier discussion of DPCM usage for Tegra and simple card driver updates. * https://lkml.org/lkml/2020/4/30/519 * https://lkml.org/lkml/2020/6/27/4 This series depends on following patch from Rob Herring: https://patchwork.ozlabs.org/project/devicetree-bindings/patch/202011170133… Changelog ========= v5 -> v6 -------- * Added ports or port description in YAML docs for Tegra AHUB devices and graph card in patch 1/6 and 2/6. Reference of audio-graph-port.yaml is used for AHUB devices. * Dropped redundant NULL check return for of_device_get_match_data() in patch 3/6. * Added 'Reviewed-by' tag from Jon Hunter. * No changes in remaining patches. v4 -> v5 -------- * Audio graph related changes were sent in separate v5 series as mentioned above and are dropped from current series. * Graph and audio graph doc patches are dropped from this series and are sent separately as mentioned above. * Minor change with phandle label for TX1 and Nano platform DT files. * No changes in other patches. v3 -> v4 -------- * Added new patches to convert graph.txt and audio-graph-card.txt to corresponding json-schema files. Later these references are used in Tegra audio graph schema. * AHUB component binding docs are updated to reflect the usage of ports/port/endpoint * More common stuff is moved into graph_parse_of() and this is used by both generic and Tegra audio graph. * DT binding for Tegra audio graph is updated to included "ports { }" * As per the suggestion 'void *data' member is dropped from 'asoc_simple_priv' and instead container method is used to maintain required custom data internal to Tegra audio graph. v2 -> v3 -------- * Dropped new compatible addition in generic graph driver after reviewing it with Morimoto-san. Instead added Tegra audio graph driver and new compatibles are added in the same. * Added new patches to expose new members for customization in audio graph driver. * Added new patch for Tegra audio graph driver and related documentation. * Minor change in below commit where mutex version of helper is used "ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM" * DT binding is updated to use the newly exposed compatibles * No changes in other patches v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. Sameer Pujar (6): ASoC: dt-bindings: tegra: Add graph bindings ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card ASoC: tegra: Add audio graph based card driver arm64: defconfig: Enable Tegra audio graph card driver arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../sound/nvidia,tegra-audio-graph-card.yaml | 187 +++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 18 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 13 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 13 +- .../bindings/sound/nvidia,tegra210-dmic.yaml | 18 +- .../bindings/sound/nvidia,tegra210-i2s.yaml | 18 +- .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 153 ++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 262 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 146 ++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_audio_graph_card.c | 251 ++++++++++++++++++++ 13 files changed, 1085 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c -- 2.7.4

3 14

Re: Steinberg UR22 Crackling and Clicks
by meschi＠posteo.de 08 Dec '20

08 Dec '20

-------- Originalnachricht -------- Betreff: Re: Steinberg UR22 Crackling and Clicks Datum: 08.12.2020 14:58 Von: meschi(a)posteo.de An: Takashi Iwai <tiwai(a)suse.de> > It's likely the issues with the implicit feedback. > > The latest development tree (either linux-next tree or > topic/usb-audio-refactoring branch of my sound.git tree) contains the > improvements of the implicit feedback mode. > > > Takashi I've compiled a kernel from git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git uname -a is "Linux meschi-Precision-WorkStation-T5500 5.10.0-rc7-next-20201208 #1 SMP PREEMPT Tue Dec 8 12:28:39 CET 2020 x86_64 x86_64 x86_64 GNU/Linux" The problem is still there. Is it possible to fallback to explicit feedback mode for the device from user space? I'd like to learn more about these feedback modes, is there a place where this is explained? I'm curious how this device is working on another machine. I'm not quite proficient with the alsa kernel subsystem to debug this myself, but I would be happy to test patches and learn along the way if somebody is interested in debugging this.

1 0

Unable to use ALSA JACK PCM plugin on Debian 10 and on raspbian
by GitHub issues - opened 08 Dec '20

08 Dec '20

alsa-project/alsa-plugins issue #15 was opened from UgoPoddine: Hello everyone, a) I have installed libasound2-plugins from repository on a Debian Buster machine (4.19.0-12-rt-amd64 #1 SMP PREEMPT RT Debian 4.19.152-1 (2020-10-18) x86_64 GNU/Linux) and on a raspberry PI3 (5.4.79-v7+ #1373 SMP Mon Nov 23 13:22:33 GMT 2020 armv7l GNU/Linux) up-to-date (none of these running pulseaudio of course) b) I have modified my .asoundrc configuration file according to https://jackaudio.org/faq/routing_alsa.html c) I can see jack and rowjack virtual devices in aplay -L d) starting qjackctl I'm unable to see the expected new capture_x devices. Of course jack runs correctly with physical devices (eg. PC mic or USB mic) e) jack is at jackd 0.125.0rc1 version f) Alsa is at Version k4.19.0-12-rt-amd64 on the AMD64 machine, at Version k5.4.79-v7+ on the RaspberriPI machine e) any attempt of playing test files over jack device (aplay -D pcm.jack <wav file> ) generates the error : ALSA lib pcm_plug.c:1286:(_snd_pcm_plug_open) Unknown field playback_ports aplay: main:828: audio open error: Invalid argument Can you help ? Thank-you Best regards Ugo Poddine Issue URL : https://github.com/alsa-project/alsa-plugins/issues/15 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: hda/realtek - Add support for Memeza EDL03 headset mic
by Chris Chiu 08 Dec '20

08 Dec '20

The Memeza laptop EDL03 with codec ALC256 can't detect the headset microphone. The headphone jack sensing works after we add a pin definition for it by ALC256_FIXUP_ASUS_MIC_NO_PRESENCE. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d5e4d0ba1008..8b9b94cfc67e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7871,6 +7871,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x115a, "Memeza EDL03", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), -- 2.20.1

2 2

[PATCH 0/7] soundwire/regmap: use _no_pm routines
by Bard Liao 08 Dec '20

08 Dec '20

When a Slave device is resumed, it may resume the bus and restart the enumeration. And Slave drivers will wait for initialization_complete complete in their resume function, however initialization_complete will complete after sdw_update_slave_status function is finished and codec driver usually call some IO functions in the update_status callback function. It will become a deadlock if we use regular read/write routines during the resuming process. This series touches both soundwire and regmap trees. commit fb5103f9d6ce ("regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ") is needed for soundwire tree to complie. On the other hands, commit 6e06a85556f9 ("soundwire: bus: add comments to explain interrupt loop filter") to commit 47b8520997a8 ("soundwire: bus: only clear valid DPN interrupts") are needed for regmap tree. Bard Liao (2): soundwire/regmap: use _no_pm functions in regmap_read/write regmap: sdw: use no_pm routines for SoundWire 1.2 MBQ Pierre-Louis Bossart (5): soundwire: bus: use sdw_update_no_pm when initializing a device soundwire: bus: use sdw_write_no_pm when setting the bus scale registers soundwire: bus: use no_pm IO routines for all interrupt handling soundwire: bus: fix confusion on device used by pm_runtime soundwire: bus: clarify dev_err/dbg device references drivers/base/regmap/regmap-sdw-mbq.c | 10 +- drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/bus.c | 135 +++++++++++++++------------ include/linux/soundwire/sdw.h | 2 + 4 files changed, 85 insertions(+), 66 deletions(-) -- 2.17.1

4 18

[PATCH 000/141] Fix fall-through warnings for Clang
by Gustavo A. R. Silva 08 Dec '20

08 Dec '20

Hi all, This series aims to fix almost all remaining fall-through warnings in order to enable -Wimplicit-fallthrough for Clang. In preparation to enable -Wimplicit-fallthrough for Clang, explicitly add multiple break/goto/return/fallthrough statements instead of just letting the code fall through to the next case. Notice that in order to enable -Wimplicit-fallthrough for Clang, this change[1] is meant to be reverted at some point. So, this patch helps to move in that direction. Something important to mention is that there is currently a discrepancy between GCC and Clang when dealing with switch fall-through to empty case statements or to cases that only contain a break/continue/return statement[2][3][4]. Now that the -Wimplicit-fallthrough option has been globally enabled[5], any compiler should really warn on missing either a fallthrough annotation or any of the other case-terminating statements (break/continue/return/ goto) when falling through to the next case statement. Making exceptions to this introduces variation in case handling which may continue to lead to bugs, misunderstandings, and a general lack of robustness. The point of enabling options like -Wimplicit-fallthrough is to prevent human error and aid developers in spotting bugs before their code is even built/ submitted/committed, therefore eliminating classes of bugs. So, in order to really accomplish this, we should, and can, move in the direction of addressing any error-prone scenarios and get rid of the unintentional fallthrough bug-class in the kernel, entirely, even if there is some minor redundancy. Better to have explicit case-ending statements than continue to have exceptions where one must guess as to the right result. The compiler will eliminate any actual redundancy. Note that there is already a patch in mainline that addresses almost 40,000 of these issues[6]. I'm happy to carry this whole series in my own tree if people are OK with it. :) [1] commit e2079e93f562c ("kbuild: Do not enable -Wimplicit-fallthrough for clang for now") [2] ClangBuiltLinux#636 [3] https://gcc.gnu.org/bugzilla/show_bug.cgi?id=91432 [4] https://godbolt.org/z/xgkvIh [5] commit a035d552a93b ("Makefile: Globally enable fall-through warning") [6] commit 4169e889e588 ("include: jhash/signal: Fix fall-through warnings for Clang") Thanks! Gustavo A. R. Silva (141): afs: Fix fall-through warnings for Clang ASoC: codecs: Fix fall-through warnings for Clang cifs: Fix fall-through warnings for Clang drm/amdgpu: Fix fall-through warnings for Clang drm/radeon: Fix fall-through warnings for Clang gfs2: Fix fall-through warnings for Clang gpio: Fix fall-through warnings for Clang IB/hfi1: Fix fall-through warnings for Clang igb: Fix fall-through warnings for Clang ima: Fix fall-through warnings for Clang ipv4: Fix fall-through warnings for Clang ixgbe: Fix fall-through warnings for Clang media: dvb-frontends: Fix fall-through warnings for Clang media: usb: dvb-usb-v2: Fix fall-through warnings for Clang netfilter: Fix fall-through warnings for Clang nfsd: Fix fall-through warnings for Clang nfs: Fix fall-through warnings for Clang qed: Fix fall-through warnings for Clang qlcnic: Fix fall-through warnings for Clang scsi: aic7xxx: Fix fall-through warnings for Clang scsi: aic94xx: Fix fall-through warnings for Clang scsi: bfa: Fix fall-through warnings for Clang staging: rtl8723bs: core: Fix fall-through warnings for Clang staging: vt6655: Fix fall-through warnings for Clang bnxt_en: Fix fall-through warnings for Clang ceph: Fix fall-through warnings for Clang drbd: Fix fall-through warnings for Clang drm/amd/display: Fix fall-through warnings for Clang e1000: Fix fall-through warnings for Clang ext2: Fix fall-through warnings for Clang ext4: Fix fall-through warnings for Clang floppy: Fix fall-through warnings for Clang fm10k: Fix fall-through warnings for Clang IB/mlx4: Fix fall-through warnings for Clang IB/qedr: Fix fall-through warnings for Clang ice: Fix fall-through warnings for Clang Input: pcspkr - Fix fall-through warnings for Clang isofs: Fix fall-through warnings for Clang ixgbevf: Fix fall-through warnings for Clang kprobes/x86: Fix fall-through warnings for Clang mm: Fix fall-through warnings for Clang net: 3c509: Fix fall-through warnings for Clang net: cassini: Fix fall-through warnings for Clang net/mlx4: Fix fall-through warnings for Clang net: mscc: ocelot: Fix fall-through warnings for Clang netxen_nic: Fix fall-through warnings for Clang nfp: Fix fall-through warnings for Clang perf/x86: Fix fall-through warnings for Clang pinctrl: Fix fall-through warnings for Clang RDMA/mlx5: Fix fall-through warnings for Clang reiserfs: Fix fall-through warnings for Clang security: keys: Fix fall-through warnings for Clang selinux: Fix fall-through warnings for Clang target: Fix fall-through warnings for Clang uprobes/x86: Fix fall-through warnings for Clang vxge: Fix fall-through warnings for Clang watchdog: Fix fall-through warnings for Clang xen-blkfront: Fix fall-through warnings for Clang regulator: as3722: Fix fall-through warnings for Clang habanalabs: Fix fall-through warnings for Clang tee: Fix fall-through warnings for Clang HID: usbhid: Fix fall-through warnings for Clang HID: input: Fix fall-through warnings for Clang ACPI: Fix fall-through warnings for Clang airo: Fix fall-through warnings for Clang ALSA: hdspm: Fix fall-through warnings for Clang ALSA: pcsp: Fix fall-through warnings for Clang ALSA: sb: Fix fall-through warnings for Clang ath5k: Fix fall-through warnings for Clang atm: fore200e: Fix fall-through warnings for Clang braille_console: Fix fall-through warnings for Clang can: peak_usb: Fix fall-through warnings for Clang carl9170: Fix fall-through warnings for Clang cfg80211: Fix fall-through warnings for Clang crypto: ccree - Fix fall-through warnings for Clang decnet: Fix fall-through warnings for Clang dm raid: Fix fall-through warnings for Clang drm/amd/pm: Fix fall-through warnings for Clang drm: Fix fall-through warnings for Clang drm/i915/gem: Fix fall-through warnings for Clang drm/nouveau/clk: Fix fall-through warnings for Clang drm/nouveau: Fix fall-through warnings for Clang drm/nouveau/therm: Fix fall-through warnings for Clang drm/via: Fix fall-through warnings for Clang firewire: core: Fix fall-through warnings for Clang hwmon: (corsair-cpro) Fix fall-through warnings for Clang hwmon: (max6621) Fix fall-through warnings for Clang i3c: master: cdns: Fix fall-through warnings for Clang ide: Fix fall-through warnings for Clang iio: adc: cpcap: Fix fall-through warnings for Clang iwlwifi: iwl-drv: Fix fall-through warnings for Clang libata: Fix fall-through warnings for Clang mac80211: Fix fall-through warnings for Clang media: atomisp: Fix fall-through warnings for Clang media: dvb_frontend: Fix fall-through warnings for Clang media: rcar_jpu: Fix fall-through warnings for Clang media: saa7134: Fix fall-through warnings for Clang mmc: sdhci-of-arasan: Fix fall-through warnings for Clang mt76: mt7615: Fix fall-through warnings for Clang mtd: cfi: Fix fall-through warnings for Clang mtd: mtdchar: Fix fall-through warnings for Clang mtd: onenand: Fix fall-through warnings for Clang mtd: rawnand: fsmc: Fix fall-through warnings for Clang mtd: rawnand: stm32_fmc2: Fix fall-through warnings for Clang net: ax25: Fix fall-through warnings for Clang net: bridge: Fix fall-through warnings for Clang net: core: Fix fall-through warnings for Clang netfilter: ipt_REJECT: Fix fall-through warnings for Clang net: netrom: Fix fall-through warnings for Clang net/packet: Fix fall-through warnings for Clang net: plip: Fix fall-through warnings for Clang net: rose: Fix fall-through warnings for Clang nl80211: Fix fall-through warnings for Clang phy: qcom-usb-hs: Fix fall-through warnings for Clang rds: Fix fall-through warnings for Clang rt2x00: Fix fall-through warnings for Clang rtl8xxxu: Fix fall-through warnings for Clang rtw88: Fix fall-through warnings for Clang rxrpc: Fix fall-through warnings for Clang scsi: aacraid: Fix fall-through warnings for Clang scsi: aha1740: Fix fall-through warnings for Clang scsi: csiostor: Fix fall-through warnings for Clang scsi: lpfc: Fix fall-through warnings for Clang scsi: stex: Fix fall-through warnings for Clang sctp: Fix fall-through warnings for Clang slimbus: messaging: Fix fall-through warnings for Clang staging: qlge: Fix fall-through warnings for Clang staging: vt6656: Fix fall-through warnings for Clang SUNRPC: Fix fall-through warnings for Clang tipc: Fix fall-through warnings for Clang tpm: Fix fall-through warnings for Clang ubi: Fix fall-through warnings for Clang usb: Fix fall-through warnings for Clang video: fbdev: lxfb_ops: Fix fall-through warnings for Clang video: fbdev: pm2fb: Fix fall-through warnings for Clang virtio_net: Fix fall-through warnings for Clang wcn36xx: Fix fall-through warnings for Clang xen/manage: Fix fall-through warnings for Clang xfrm: Fix fall-through warnings for Clang zd1201: Fix fall-through warnings for Clang Input: libps2 - Fix fall-through warnings for Clang arch/x86/events/core.c | 2 +- arch/x86/kernel/kprobes/core.c | 1 + arch/x86/kernel/uprobes.c | 2 ++ drivers/accessibility/braille/braille_console.c | 1 + drivers/acpi/sbshc.c | 1 + drivers/ata/libata-eh.c | 1 + drivers/atm/fore200e.c | 1 + drivers/block/drbd/drbd_receiver.c | 1 + drivers/block/drbd/drbd_req.c | 1 + drivers/block/floppy.c | 1 + drivers/block/xen-blkfront.c | 1 + drivers/char/tpm/eventlog/tpm1.c | 1 + drivers/crypto/ccree/cc_cipher.c | 3 +++ drivers/firewire/core-topology.c | 1 + drivers/gpio/gpio-ath79.c | 1 + drivers/gpio/gpiolib-acpi.c | 1 + drivers/gpu/drm/amd/amdgpu/gfx_v10_0.c | 1 + drivers/gpu/drm/amd/amdgpu/gfx_v9_0.c | 1 + drivers/gpu/drm/amd/amdgpu/gmc_v9_0.c | 1 + drivers/gpu/drm/amd/amdgpu/vi.c | 1 + drivers/gpu/drm/amd/display/dc/bios/bios_parser.c | 1 + drivers/gpu/drm/amd/display/dc/bios/bios_parser2.c | 2 ++ drivers/gpu/drm/amd/display/dc/core/dc_link.c | 1 + drivers/gpu/drm/amd/pm/powerplay/si_dpm.c | 2 +- .../gpu/drm/amd/pm/powerplay/smumgr/polaris10_smumgr.c | 1 + drivers/gpu/drm/drm_bufs.c | 1 + drivers/gpu/drm/i915/gem/i915_gem_shrinker.c | 1 + drivers/gpu/drm/nouveau/nouveau_bo.c | 1 + drivers/gpu/drm/nouveau/nouveau_connector.c | 1 + drivers/gpu/drm/nouveau/nvkm/subdev/clk/nv50.c | 1 + drivers/gpu/drm/nouveau/nvkm/subdev/therm/gf119.c | 1 + drivers/gpu/drm/radeon/ci_dpm.c | 2 +- drivers/gpu/drm/radeon/r300.c | 1 + drivers/gpu/drm/radeon/si_dpm.c | 2 +- drivers/gpu/drm/via/via_irq.c | 1 + drivers/hid/hid-input.c | 1 + drivers/hid/usbhid/hid-core.c | 2 ++ drivers/hwmon/corsair-cpro.c | 1 + drivers/hwmon/max6621.c | 2 +- drivers/i3c/master/i3c-master-cdns.c | 2 ++ drivers/ide/siimage.c | 1 + drivers/iio/adc/cpcap-adc.c | 1 + drivers/infiniband/hw/hfi1/qp.c | 1 + drivers/infiniband/hw/hfi1/tid_rdma.c | 5 +++++ drivers/infiniband/hw/mlx4/mad.c | 1 + drivers/infiniband/hw/mlx5/qp.c | 1 + drivers/infiniband/hw/qedr/main.c | 1 + drivers/input/misc/pcspkr.c | 1 + drivers/input/serio/libps2.c | 2 +- drivers/md/dm-raid.c | 1 + drivers/media/dvb-core/dvb_frontend.c | 1 + drivers/media/dvb-frontends/cx24120.c | 1 + drivers/media/dvb-frontends/dib0090.c | 2 ++ drivers/media/dvb-frontends/drxk_hard.c | 1 + drivers/media/dvb-frontends/m88rs2000.c | 1 + drivers/media/pci/saa7134/saa7134-tvaudio.c | 1 + drivers/media/platform/rcar_jpu.c | 1 + drivers/media/usb/dvb-usb-v2/af9015.c | 1 + drivers/media/usb/dvb-usb-v2/lmedm04.c | 1 + drivers/misc/habanalabs/gaudi/gaudi.c | 1 + drivers/mmc/host/sdhci-of-arasan.c | 4 ++++ drivers/mtd/chips/cfi_cmdset_0001.c | 1 + drivers/mtd/chips/cfi_cmdset_0002.c | 2 ++ drivers/mtd/chips/cfi_cmdset_0020.c | 2 ++ drivers/mtd/mtdchar.c | 1 + drivers/mtd/nand/onenand/onenand_samsung.c | 1 + drivers/mtd/nand/raw/fsmc_nand.c | 1 + drivers/mtd/nand/raw/stm32_fmc2_nand.c | 2 ++ drivers/mtd/ubi/build.c | 1 + drivers/net/can/usb/peak_usb/pcan_usb_core.c | 2 ++ drivers/net/ethernet/3com/3c509.c | 1 + drivers/net/ethernet/broadcom/bnxt/bnxt.c | 1 + drivers/net/ethernet/intel/e1000/e1000_hw.c | 1 + drivers/net/ethernet/intel/fm10k/fm10k_mbx.c | 2 ++ drivers/net/ethernet/intel/ice/ice_txrx_lib.c | 1 + drivers/net/ethernet/intel/igb/e1000_phy.c | 1 + drivers/net/ethernet/intel/igb/igb_ethtool.c | 1 + drivers/net/ethernet/intel/igb/igb_ptp.c | 1 + drivers/net/ethernet/intel/ixgbe/ixgbe_82599.c | 2 ++ drivers/net/ethernet/intel/ixgbe/ixgbe_common.c | 1 + drivers/net/ethernet/intel/ixgbe/ixgbe_lib.c | 1 + drivers/net/ethernet/intel/ixgbe/ixgbe_ptp.c | 1 + drivers/net/ethernet/intel/ixgbevf/ixgbevf_main.c | 1 + drivers/net/ethernet/mellanox/mlx4/resource_tracker.c | 1 + drivers/net/ethernet/mscc/ocelot_vcap.c | 1 + drivers/net/ethernet/neterion/vxge/vxge-config.c | 1 + drivers/net/ethernet/netronome/nfp/nfp_net_repr.c | 1 + drivers/net/ethernet/qlogic/netxen/netxen_nic_init.c | 1 + drivers/net/ethernet/qlogic/qed/qed_l2.c | 1 + drivers/net/ethernet/qlogic/qed/qed_sriov.c | 1 + drivers/net/ethernet/qlogic/qlcnic/qlcnic_io.c | 1 + drivers/net/ethernet/qlogic/qlcnic/qlcnic_main.c | 1 + drivers/net/ethernet/sun/cassini.c | 1 + drivers/net/plip/plip.c | 2 ++ drivers/net/virtio_net.c | 1 + drivers/net/wireless/ath/ath5k/mac80211-ops.c | 1 + drivers/net/wireless/ath/carl9170/tx.c | 1 + drivers/net/wireless/ath/wcn36xx/smd.c | 2 +- drivers/net/wireless/cisco/airo.c | 1 + drivers/net/wireless/intel/iwlwifi/iwl-drv.c | 2 +- drivers/net/wireless/mediatek/mt76/mt7615/eeprom.c | 2 +- drivers/net/wireless/ralink/rt2x00/rt2x00queue.c | 1 + drivers/net/wireless/realtek/rtl8xxxu/rtl8xxxu_core.c | 8 ++++---- drivers/net/wireless/realtek/rtw88/fw.c | 2 +- drivers/net/wireless/zydas/zd1201.c | 2 +- drivers/phy/qualcomm/phy-qcom-usb-hs.c | 1 + drivers/pinctrl/renesas/pinctrl-rza1.c | 1 + drivers/regulator/as3722-regulator.c | 3 ++- drivers/scsi/aacraid/commsup.c | 1 + drivers/scsi/aha1740.c | 1 + drivers/scsi/aic7xxx/aic79xx_core.c | 4 +++- drivers/scsi/aic7xxx/aic7xxx_core.c | 4 ++-- drivers/scsi/aic94xx/aic94xx_scb.c | 2 ++ drivers/scsi/aic94xx/aic94xx_task.c | 2 ++ drivers/scsi/bfa/bfa_fcs_lport.c | 2 +- drivers/scsi/bfa/bfa_ioc.c | 6 ++++-- drivers/scsi/csiostor/csio_wr.c | 1 + drivers/scsi/lpfc/lpfc_bsg.c | 1 + drivers/scsi/stex.c | 1 + drivers/slimbus/messaging.c | 1 + drivers/staging/media/atomisp/pci/runtime/isys/src/rx.c | 1 + drivers/staging/qlge/qlge_main.c | 1 + drivers/staging/rtl8723bs/core/rtw_cmd.c | 1 + drivers/staging/rtl8723bs/core/rtw_mlme_ext.c | 1 + drivers/staging/rtl8723bs/core/rtw_wlan_util.c | 1 + drivers/staging/vt6655/device_main.c | 1 + drivers/staging/vt6655/rxtx.c | 2 ++ drivers/staging/vt6656/main_usb.c | 1 + drivers/target/target_core_iblock.c | 1 + drivers/target/target_core_pr.c | 1 + drivers/tee/tee_core.c | 1 + drivers/usb/gadget/function/f_fs.c | 2 ++ drivers/usb/gadget/function/f_loopback.c | 2 +- drivers/usb/gadget/function/f_sourcesink.c | 1 + drivers/usb/gadget/udc/dummy_hcd.c | 2 ++ drivers/usb/host/fotg210-hcd.c | 2 +- drivers/usb/host/isp116x-hcd.c | 1 + drivers/usb/host/max3421-hcd.c | 1 + drivers/usb/host/oxu210hp-hcd.c | 1 + drivers/usb/misc/yurex.c | 1 + drivers/usb/musb/tusb6010.c | 1 + drivers/usb/storage/ene_ub6250.c | 1 + drivers/usb/storage/uas.c | 1 + drivers/video/fbdev/geode/lxfb_ops.c | 1 + drivers/video/fbdev/pm2fb.c | 1 + drivers/watchdog/machzwd.c | 1 + drivers/xen/manage.c | 1 + fs/afs/cmservice.c | 5 +++++ fs/afs/fsclient.c | 4 ++++ fs/afs/vlclient.c | 1 + fs/ceph/dir.c | 2 ++ fs/cifs/inode.c | 1 + fs/cifs/sess.c | 1 + fs/cifs/smbdirect.c | 1 + fs/ext2/inode.c | 1 + fs/ext4/super.c | 1 + fs/gfs2/inode.c | 2 ++ fs/gfs2/recovery.c | 1 + fs/isofs/rock.c | 1 + fs/nfs/nfs3acl.c | 1 + fs/nfs/nfs4client.c | 1 + fs/nfs/nfs4proc.c | 2 ++ fs/nfs/nfs4state.c | 1 + fs/nfs/pnfs.c | 2 ++ fs/nfsd/nfs4state.c | 1 + fs/nfsd/nfsctl.c | 1 + fs/reiserfs/namei.c | 1 + mm/mm_init.c | 1 + mm/vmscan.c | 1 + net/ax25/af_ax25.c | 1 + net/bridge/br_input.c | 1 + net/core/dev.c | 1 + net/decnet/dn_route.c | 2 +- net/ipv4/ah4.c | 1 + net/ipv4/esp4.c | 1 + net/ipv4/fib_semantics.c | 1 + net/ipv4/ip_vti.c | 1 + net/ipv4/ipcomp.c | 1 + net/ipv4/netfilter/ipt_REJECT.c | 1 + net/mac80211/cfg.c | 2 ++ net/netfilter/nf_conntrack_proto_dccp.c | 1 + net/netfilter/nf_tables_api.c | 1 + net/netfilter/nft_ct.c | 1 + net/netrom/nr_route.c | 4 ++++ net/packet/af_packet.c | 1 + net/rds/tcp_connect.c | 1 + net/rds/threads.c | 2 ++ net/rose/rose_route.c | 2 ++ net/rxrpc/af_rxrpc.c | 1 + net/sctp/input.c | 3 ++- net/sunrpc/rpc_pipe.c | 1 + net/sunrpc/xprtsock.c | 1 + net/tipc/link.c | 1 + net/wireless/nl80211.c | 1 + net/wireless/util.c | 1 + net/xfrm/xfrm_interface.c | 1 + security/integrity/ima/ima_main.c | 1 + security/integrity/ima/ima_policy.c | 2 ++ security/keys/process_keys.c | 1 + security/selinux/hooks.c | 1 + sound/drivers/pcsp/pcsp_input.c | 1 + sound/isa/sb/sb8_main.c | 1 + sound/pci/rme9652/hdspm.c | 1 + sound/soc/codecs/adav80x.c | 1 + sound/soc/codecs/arizona.c | 1 + sound/soc/codecs/cs42l52.c | 1 + sound/soc/codecs/cs42l56.c | 1 + sound/soc/codecs/cs47l92.c | 1 + sound/soc/codecs/wm8962.c | 1 + 209 files changed, 264 insertions(+), 26 deletions(-) -- 2.27.0

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[PATCH 1/1] dt-bindings: mediatek: mt8192: eliminate yamllint warnings
by Zhen Lei 08 Dec '20

08 Dec '20

Eliminate the following yamllint warnings: ./Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml :10:4: [warning] wrong indentation: expected 2 but found 3 (indentation) Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- .../devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index bf8c8ba25009dcc..54650823b29a41d 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Mediatek MT8192 with MT6359, RT1015 and RT5682 ASoC sound card driver maintainers: - - Jiaxin Yu <jiaxin.yu(a)mediatek.com> - - Shane Chien <shane.chien(a)mediatek.com> + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> description: This binding describes the MT8192 sound card. -- 1.8.3

2 1

[PATCH] ASoC: pcm: send DAPM_STREAM_STOP event in dpcm_fe_dai_shutdown
by Ranjani Sridharan 07 Dec '20

07 Dec '20

A recent change removed the call to send the DAPM_STREAM_STOP event in dpcm_fe_dai_shutdown. But this causes a regression when a PCM prepare is not paired with a hw_free. So, add the DAPM_STREAM_STOP event back to dpcm_fe_dai_shutdown() to fix this. The new sequence would be: soc_pcm_prepare() -> SND_SOC_DAPM_STREAM_START soc_pcm_hw_free() -> soc_pcm_hw_free() -> SND_SOC_DAPM_STREAM_STOP dpcm_fe_dai_shutdown() -> SND_SOC_DAPM_STREAM_STOP Note that the DAPM_STREAM_STOP will be called twice but it seems harmless. Fixes: a27b421f1d04 ('ASoC: pcm: call snd_soc_dapm_stream_stop() in soc_pcm_hw_clean') Reported-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5250124dc8f5..ae062e4d1ce8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1860,6 +1860,9 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) /* now shutdown the frontend */ soc_pcm_close(substream); + /* run the stream stop event */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; -- 2.25.1

3 3

[PATCH] ASoC: SOF: trace: Add runtime trace filtering mechanism
by Kai Vehmanen 07 Dec '20

07 Dec '20

From: Karol Trzcinski <karolx.trzcinski(a)linux.intel.com> The "filter" debugfs file defines the log levels used by the firmware and reported by sof-logger. The file contains the formatted entry list, where each entry follows the following syntax in plain text: log_level uuid_id pipe_id comp_id; This file may be updated by userspace applications such sof-logger, or directly by the user during debugging process. An unused (wildcard) pipe_id or comp_id value should be set to -1, uuid_id is hexadecimal value, so when unused then should be set to 0. When the file is modified, an IPC command is sent to FW with new trace levels for selected components in filter elements list. Signed-off-by: Karol Trzcinski <karolx.trzcinski(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/sof/header.h | 1 + include/sound/sof/trace.h | 28 +++++ sound/soc/sof/trace.c | 224 +++++++++++++++++++++++++++++++++++++ 3 files changed, 253 insertions(+) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index c93f08334bbe..4c747c52e01b 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -118,6 +118,7 @@ #define SOF_IPC_TRACE_DMA_PARAMS SOF_CMD_TYPE(0x001) #define SOF_IPC_TRACE_DMA_POSITION SOF_CMD_TYPE(0x002) #define SOF_IPC_TRACE_DMA_PARAMS_EXT SOF_CMD_TYPE(0x003) +#define SOF_IPC_TRACE_FILTER_UPDATE SOF_CMD_TYPE(0x004) /**< ABI3.17 */ /* debug */ #define SOF_IPC_DEBUG_MEM_USAGE SOF_CMD_TYPE(0x001) diff --git a/include/sound/sof/trace.h b/include/sound/sof/trace.h index c31a94a13ce0..25ea99f62d37 100644 --- a/include/sound/sof/trace.h +++ b/include/sound/sof/trace.h @@ -43,6 +43,34 @@ struct sof_ipc_dma_trace_posn { uint32_t messages; /* total trace messages */ } __packed; +/* Values used in sof_ipc_trace_filter_elem: */ + +/* bits 6..0 */ +#define SOF_IPC_TRACE_FILTER_ELEM_SET_LEVEL 0x01 /**< trace level for selected components */ +#define SOF_IPC_TRACE_FILTER_ELEM_BY_UUID 0x02 /**< filter by uuid key */ +#define SOF_IPC_TRACE_FILTER_ELEM_BY_PIPE 0x03 /**< filter by pipeline */ +#define SOF_IPC_TRACE_FILTER_ELEM_BY_COMP 0x04 /**< filter by component id */ + +/* bit 7 */ +#define SOF_IPC_TRACE_FILTER_ELEM_FIN 0x80 /**< mark last filter in set */ + +/* bits 31..8: Unused */ + +/** part of sof_ipc_trace_filter, ABI3.17 */ +struct sof_ipc_trace_filter_elem { + uint32_t key; /**< SOF_IPC_TRACE_FILTER_ELEM_ {LEVEL, UUID, COMP, PIPE} */ + uint32_t value; /**< element value */ +} __packed; + +/** Runtime tracing filtration data - SOF_IPC_TRACE_FILTER_UPDATE, ABI3.17 */ +struct sof_ipc_trace_filter { + struct sof_ipc_cmd_hdr hdr; /**< IPC command header */ + uint32_t elem_cnt; /**< number of entries in elems[] array */ + uint32_t reserved[8]; /**< reserved for future usage */ + /** variable size array with new filtering settings */ + struct sof_ipc_trace_filter_elem elems[]; +} __packed; + /* * Commom debug */ diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index 69889241a092..f72a6e83e6af 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -13,6 +13,225 @@ #include "sof-priv.h" #include "ops.h" +#define TRACE_FILTER_ELEMENTS_PER_ENTRY 4 +#define TRACE_FILTER_MAX_CONFIG_STRING_LENGTH 1024 + +static int trace_filter_append_elem(struct snd_sof_dev *sdev, uint32_t key, uint32_t value, + struct sof_ipc_trace_filter_elem *elem_list, + int capacity, int *counter) +{ + if (*counter >= capacity) + return -ENOMEM; + + elem_list[*counter].key = key; + elem_list[*counter].value = value; + ++*counter; + + return 0; +} + +static int trace_filter_parse_entry(struct snd_sof_dev *sdev, const char *line, + struct sof_ipc_trace_filter_elem *elem, + int capacity, int *counter) +{ + int len = strlen(line); + int cnt = *counter; + uint32_t uuid_id; + int log_level; + int pipe_id; + int comp_id; + int read; + int ret; + + /* ignore empty content */ + ret = sscanf(line, " %n", &read); + if (!ret && read == len) + return len; + + ret = sscanf(line, " %d %x %d %d %n", &log_level, &uuid_id, &pipe_id, &comp_id, &read); + if (ret != TRACE_FILTER_ELEMENTS_PER_ENTRY || read != len) { + dev_err(sdev->dev, "error: invalid trace filter entry '%s'\n", line); + return -EINVAL; + } + + if (uuid_id > 0) { + ret = trace_filter_append_elem(sdev, SOF_IPC_TRACE_FILTER_ELEM_BY_UUID, + uuid_id, elem, capacity, &cnt); + if (ret) + return ret; + } + if (pipe_id >= 0) { + ret = trace_filter_append_elem(sdev, SOF_IPC_TRACE_FILTER_ELEM_BY_PIPE, + pipe_id, elem, capacity, &cnt); + if (ret) + return ret; + } + if (comp_id >= 0) { + ret = trace_filter_append_elem(sdev, SOF_IPC_TRACE_FILTER_ELEM_BY_COMP, + comp_id, elem, capacity, &cnt); + if (ret) + return ret; + } + + ret = trace_filter_append_elem(sdev, SOF_IPC_TRACE_FILTER_ELEM_SET_LEVEL | + SOF_IPC_TRACE_FILTER_ELEM_FIN, + log_level, elem, capacity, &cnt); + if (ret) + return ret; + + /* update counter only when parsing whole entry passed */ + *counter = cnt; + + return len; +} + +static int trace_filter_parse(struct snd_sof_dev *sdev, char *string, + int *out_elem_cnt, + struct sof_ipc_trace_filter_elem **out) +{ + static const char entry_delimiter[] = ";"; + char *entry = string; + int capacity = 0; + int entry_len; + int cnt = 0; + + /* + * Each entry contains at least 1, up to TRACE_FILTER_ELEMENTS_PER_ENTRY + * IPC elements, depending on content. Calculate IPC elements capacity + * for the input string where each element is set. + */ + while (entry) { + capacity += TRACE_FILTER_ELEMENTS_PER_ENTRY; + entry = strchr(entry + 1, entry_delimiter[0]); + } + *out = kmalloc(capacity * sizeof(**out), GFP_KERNEL); + if (!*out) + return -ENOMEM; + + /* split input string by ';', and parse each entry separately in trace_filter_parse_entry */ + while ((entry = strsep(&string, entry_delimiter))) { + entry_len = trace_filter_parse_entry(sdev, entry, *out, capacity, &cnt); + if (entry_len < 0) { + dev_err(sdev->dev, "error: %s failed for '%s', %d\n", __func__, entry, + entry_len); + return -EINVAL; + } + } + + *out_elem_cnt = cnt; + + return 0; +} + +static int sof_ipc_trace_update_filter(struct snd_sof_dev *sdev, int num_elems, + struct sof_ipc_trace_filter_elem *elems) +{ + struct sof_ipc_trace_filter *msg; + struct sof_ipc_reply reply; + size_t size; + int ret; + + size = struct_size(msg, elems, num_elems); + if (size > SOF_IPC_MSG_MAX_SIZE) + return -EINVAL; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + + msg->hdr.size = size; + msg->hdr.cmd = SOF_IPC_GLB_TRACE_MSG | SOF_IPC_TRACE_FILTER_UPDATE; + msg->elem_cnt = num_elems; + memcpy(&msg->elems[0], elems, num_elems * sizeof(*elems)); + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(sdev->dev); + dev_err(sdev->dev, "error: enabling device failed: %d\n", ret); + goto error; + } + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + pm_runtime_mark_last_busy(sdev->dev); + pm_runtime_put_autosuspend(sdev->dev); + +error: + kfree(msg); + return ret ? ret : reply.error; +} + +static ssize_t sof_dfsentry_trace_filter_write(struct file *file, const char __user *from, + size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct sof_ipc_trace_filter_elem *elems = NULL; + struct snd_sof_dev *sdev = dfse->sdev; + loff_t pos = 0; + int num_elems; + char *string; + int ret; + + if (count > TRACE_FILTER_MAX_CONFIG_STRING_LENGTH) { + dev_err(sdev->dev, "%s too long input, %zu > %d\n", __func__, count, + TRACE_FILTER_MAX_CONFIG_STRING_LENGTH); + return -EINVAL; + } + + string = kmalloc(count + 1, GFP_KERNEL); + if (!string) + return -ENOMEM; + + /* assert null termination */ + string[count] = 0; + ret = simple_write_to_buffer(string, count, &pos, from, count); + if (ret < 0) + goto error; + + ret = trace_filter_parse(sdev, string, &num_elems, &elems); + if (ret < 0) { + dev_err(sdev->dev, "error: fail in trace_filter_parse, %d\n", ret); + goto error; + } + + if (num_elems) { + ret = sof_ipc_trace_update_filter(sdev, num_elems, elems); + if (ret < 0) { + dev_err(sdev->dev, "error: fail in sof_ipc_trace_update_filter %d\n", ret); + goto error; + } + } + ret = count; +error: + kfree(string); + kfree(elems); + return ret; +} + +static const struct file_operations sof_dfs_trace_filter_fops = { + .open = simple_open, + .write = sof_dfsentry_trace_filter_write, + .llseek = default_llseek, +}; + +static int trace_debugfs_filter_create(struct snd_sof_dev *sdev) +{ + struct snd_sof_dfsentry *dfse; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->sdev = sdev; + dfse->type = SOF_DFSENTRY_TYPE_BUF; + + debugfs_create_file("filter", 0200, sdev->debugfs_root, dfse, + &sof_dfs_trace_filter_fops); + /* add to dfsentry list */ + list_add(&dfse->list, &sdev->dfsentry_list); + + return 0; +} + static size_t sof_trace_avail(struct snd_sof_dev *sdev, loff_t pos, size_t buffer_size) { @@ -135,10 +354,15 @@ static const struct file_operations sof_dfs_trace_fops = { static int trace_debugfs_create(struct snd_sof_dev *sdev) { struct snd_sof_dfsentry *dfse; + int ret; if (!sdev) return -EINVAL; + ret = trace_debugfs_filter_create(sdev); + if (ret < 0) + dev_err(sdev->dev, "error: fail in %s, %d", __func__, ret); + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); if (!dfse) return -ENOMEM; -- 2.28.0

2 1

Alsa-lib compilation error
by Caio Lima 07 Dec '20

07 Dec '20

Hello ALSA-devel team, nice to contact you. Firstly I want to let you know I'm a fan of the work of you guys of the ALSA Project, specifically of the alsa-lib of which I had a little experience. I'm needing help with the compile of the alsa-lib library, I'm using cygwin and following the instructions in the "INSTALL" file inside the source but during the compiling with the command "./gitcompile" I'm getting some errors, here the entire log: Caio@Caio-PC ~/alsa-lib $ ./gitcompile configure.ac:30: installing './compile' configure.ac:15: installing './config.guess' configure.ac:15: installing './config.sub' configure.ac:16: installing './install-sh' configure.ac:16: installing './missing' alsalisp/Makefile.am: installing './depcomp' parallel-tests: installing './test-driver' CFLAGS=-O2 -Wall -W -Wunused-const-variable=0 -pipe -g ./configure --disable-aload --prefix=/usr --libdir=/usr/lib --with-plugindir=/usr/lib/alsa-lib --with-pkgconfdir=/usr/lib/pkgconfig configure: loading site script /usr/share/config.site checking build system type... x86_64-pc-cygwin checking host system type... x86_64-pc-cygwin checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... /usr/bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether make supports nested variables... yes checking whether to enable maintainer-specific portions of Makefiles... yes checking whether make supports nested variables... (cached) yes checking for gcc... gcc checking whether the C compiler works... yes checking for C compiler default output file name... a.exe checking for suffix of executables... .exe checking whether we are cross compiling... no checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking whether gcc understands -c and -o together... yes checking whether make supports the include directive... yes (GNU style) checking dependency style of gcc... gcc3 checking how to run the C preprocessor... gcc -E checking for grep that handles long lines and -e... /usr/bin/grep checking for egrep... /usr/bin/grep -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking minix/config.h usability... no checking minix/config.h presence... no checking for minix/config.h... no checking whether it is safe to define __EXTENSIONS__... yes checking whether ln -s works... yes checking how to print strings... printf checking for a sed that does not truncate output... /usr/bin/sed checking for fgrep... /usr/bin/grep -F checking for ld used by gcc... /usr/x86_64-pc-cygwin/bin/ld.exe checking if the linker (/usr/x86_64-pc-cygwin/bin/ld.exe) is GNU ld... yes checking for BSD- or MS-compatible name lister (nm)... /usr/bin/nm -B checking the name lister (/usr/bin/nm -B) interface... BSD nm checking the maximum length of command line arguments... 8192 checking how to convert x86_64-pc-cygwin file names to x86_64-pc-cygwin format... func_convert_file_noop checking how to convert x86_64-pc-cygwin file names to toolchain format... func_convert_file_noop checking for /usr/x86_64-pc-cygwin/bin/ld.exe option to reload object files... -r checking for objdump... objdump checking how to recognize dependent libraries... file_magic ^x86 archive import|^x86 DLL checking for dlltool... dlltool checking how to associate runtime and link libraries... func_cygming_dll_for_implib checking for ar... ar checking for archiver @FILE support... @ checking for strip... strip checking for ranlib... ranlib checking command to parse /usr/bin/nm -B output from gcc object... ok checking for sysroot... no checking for a working dd... /usr/bin/dd checking how to truncate binary pipes... /usr/bin/dd bs=4096 count=1 checking for mt... no checking if : is a manifest tool... no checking for dlfcn.h... yes checking for objdir... .libs checking if gcc supports -fno-rtti -fno-exceptions... no checking for gcc option to produce PIC... -DDLL_EXPORT -DPIC checking if gcc PIC flag -DDLL_EXPORT -DPIC works... yes checking if gcc static flag -static works... yes checking if gcc supports -c -o file.o... yes checking if gcc supports -c -o file.o... (cached) yes checking whether the gcc linker (/usr/x86_64-pc-cygwin/bin/ld.exe) supports shared libraries... yes checking whether -lc should be explicitly linked in... yes checking dynamic linker characteristics... Win32 ld.exe checking how to hardcode library paths into programs... immediate checking whether a program can dlopen itself... yes checking whether a statically linked program can dlopen itself... yes checking whether stripping libraries is possible... yes checking if libtool supports shared libraries... yes checking whether to build shared libraries... yes checking whether to build static libraries... no checking if gcc supports -Wl,--no-undefined flag... yes checking if gcc supports -Wl,-z,defs flag... checking for ANSI C header files... (cached) yes checking for an ANSI C-conforming const... yes checking for inline... inline checking whether time.h and sys/time.h may both be included... yes checking whether gcc needs -traditional... no checking for uselocale... yes checking for library version... major 1 minor 2 subminor 4 extrastr extraver 1000000 checking for versioned symbols... yes checking for symbolic-functions... no checking for custom symbol prefixes... checking for debug... yes checking for debug assert... no checking for tmpdir... /tmp checking for softfloat... no checking for libdl... checking for dlsym in -ldl... yes checking for pthread... checking for pthread_join in -lpthread... yes checking whether PTHREAD_MUTEX_RECURSIVE is declared... yes checking for __thread... yes checking for librt... checking for clock_gettime in -lrt... yes checking for use of wordexp... no checking endian.h usability... yes checking endian.h presence... yes checking for endian.h... yes checking sys/endian.h usability... no checking sys/endian.h presence... no checking for sys/endian.h... no checking sys/shm.h usability... yes checking sys/shm.h presence... yes checking for sys/shm.h... yes checking for resmgr support... no checking for aload* support... no checking for ALSA device file directory... /dev/snd/ checking for aload* device file directory... /dev/ checking whether GCC supports builtin atomic intrinsics... yes checking for default lockless dmix... no checking for max number of cards... 32 checking for thread-safe API functions... yes Making a symlink include/alsa checking that generated files are newer than configure... done configure: creating ./config.status config.status: creating Makefile config.status: creating doc/Makefile config.status: creating doc/pictures/Makefile config.status: creating doc/doxygen.cfg config.status: creating include/Makefile config.status: creating include/sound/Makefile config.status: creating include/sound/uapi/Makefile config.status: creating src/Versions config.status: creating src/Makefile config.status: creating src/control/Makefile config.status: creating src/mixer/Makefile config.status: creating src/pcm/Makefile config.status: creating src/pcm/scopes/Makefile config.status: creating src/rawmidi/Makefile config.status: creating src/timer/Makefile config.status: creating src/hwdep/Makefile config.status: creating src/seq/Makefile config.status: creating src/ucm/Makefile config.status: creating src/alisp/Makefile config.status: creating src/topology/Makefile config.status: creating src/conf/Makefile config.status: creating src/conf/cards/Makefile config.status: creating src/conf/pcm/Makefile config.status: creating modules/Makefile config.status: creating modules/mixer/Makefile config.status: creating modules/mixer/simple/Makefile config.status: creating alsalisp/Makefile config.status: creating aserver/Makefile config.status: creating test/Makefile config.status: creating test/lsb/Makefile config.status: creating utils/Makefile config.status: creating utils/alsa-lib.spec config.status: creating utils/alsa.pc config.status: creating utils/alsa-topology.pc config.status: creating include/config.h config.status: executing depfiles commands config.status: executing libtool commands Creating asoundlib.h... Making all in doc make[1]: Entering directory '/home/Caio/alsa-lib/doc' Making all in pictures make[2]: Entering directory '/home/Caio/alsa-lib/doc/pictures' make[2]: Nothing to be done for 'all'. make[2]: Leaving directory '/home/Caio/alsa-lib/doc/pictures' make[2]: Entering directory '/home/Caio/alsa-lib/doc' make[2]: Nothing to be done for 'all-am'. make[2]: Leaving directory '/home/Caio/alsa-lib/doc' make[1]: Leaving directory '/home/Caio/alsa-lib/doc' Making all in include make[1]: Entering directory '/home/Caio/alsa-lib/include' make all-recursive make[2]: Entering directory '/home/Caio/alsa-lib/include' Making all in sound make[3]: Entering directory '/home/Caio/alsa-lib/include/sound' Making all in uapi make[4]: Entering directory '/home/Caio/alsa-lib/include/sound/uapi' make[4]: Nothing to be done for 'all'. make[4]: Leaving directory '/home/Caio/alsa-lib/include/sound/uapi' make[4]: Entering directory '/home/Caio/alsa-lib/include/sound' make[4]: Nothing to be done for 'all-am'. make[4]: Leaving directory '/home/Caio/alsa-lib/include/sound' make[3]: Leaving directory '/home/Caio/alsa-lib/include/sound' make[3]: Entering directory '/home/Caio/alsa-lib/include' Updating version.h make[3]: Leaving directory '/home/Caio/alsa-lib/include' make[2]: Leaving directory '/home/Caio/alsa-lib/include' make[1]: Leaving directory '/home/Caio/alsa-lib/include' Making all in src make[1]: Entering directory '/home/Caio/alsa-lib/src' Making all in control make[2]: Entering directory '/home/Caio/alsa-lib/src/control' CC cards.lo In file included from ../../include/local.h:25, from control_local.h:22, from cards.c:35: ../../include/config.h:226: warning: "__SYMBOL_PREFIX" redefined 226 | #define __SYMBOL_PREFIX "" | In file included from /usr/include/sys/config.h:238, from /usr/include/_ansi.h:11, from /usr/include/stdio.h:29, from cards.c:28: /usr/include/cygwin/config.h:58: note: this is the location of the previous definition 58 | # define __SYMBOL_PREFIX | In file included from ../../include/sound/asound.h:11, from ../../include/local.h:146, from control_local.h:22, from cards.c:35: ../../include/alsa/sound/uapi/asound.h:512:2: error: unknown type name ‘__s64’ 512 | __s64 tv_sec; | ^~~~~ ../../include/alsa/sound/uapi/asound.h:513:2: error: unknown type name ‘__s64’ 513 | __s64 tv_nsec; | ^~~~~ cards.c: In function ‘snd_card_load2’: ../../include/alsa/sound/uapi/asound.h:1064:35: warning: implicit declaration of function ‘_IOR’ [-Wimplicit-function-declaration] 1064 | #define SNDRV_CTL_IOCTL_CARD_INFO _IOR('U', 0x01, struct snd_ctl_card_info) | ^~~~ cards.c:49:23: note: in expansion of macro ‘SNDRV_CTL_IOCTL_CARD_INFO’ 49 | if (ioctl(open_dev, SNDRV_CTL_IOCTL_CARD_INFO, &info) < 0) { | ^~~~~~~~~~~~~~~~~~~~~~~~~ ../../include/alsa/sound/uapi/asound.h:1064:51: error: expected expression before ‘struct’ 1064 | #define SNDRV_CTL_IOCTL_CARD_INFO _IOR('U', 0x01, struct snd_ctl_card_info) | ^~~~~~ cards.c:49:23: note: in expansion of macro ‘SNDRV_CTL_IOCTL_CARD_INFO’ 49 | if (ioctl(open_dev, SNDRV_CTL_IOCTL_CARD_INFO, &info) < 0) { | ^~~~~~~~~~~~~~~~~~~~~~~~~ In file included from cards.c:32: cards.c: In function ‘snd_card_get_index’: cards.c:133:15: warning: array subscript has type ‘char’ [-Wchar-subscripts] 133 | if ((isdigit(*string) && *(string + 1) == 0) || | ^~~~~~~ cards.c:134:15: warning: array subscript has type ‘char’ [-Wchar-subscripts] 134 | (isdigit(*string) && isdigit(*(string + 1)) && *(string + 2) == 0)) { | ^~~~~~~ cards.c:134:35: warning: array subscript has type ‘char’ [-Wchar-subscripts] 134 | (isdigit(*string) && isdigit(*(string + 1)) && *(string + 2) == 0)) { | ^~~~~~~~~~~~~ make[2]: *** [Makefile:404: cards.lo] Error 1 make[2]: Leaving directory '/home/Caio/alsa-lib/src/control' make[1]: *** [Makefile:554: all-recursive] Error 1 make[1]: Leaving directory '/home/Caio/alsa-lib/src' make: *** [Makefile:405: all-recursive] Error 1 I would really appreciate your help with this issue! Regards Caio H. Lima

1 0

[RFC SOF 0/2] topology: Try to add sof-cht-nau8824 topology file
by Hans de Goede 07 Dec '20

07 Dec '20

Hi All, I have veen working on making the cht_bsw_nau8824 machine driver work together with the sof-audio-acpi driver. The sof topology files are missing a .tplg file for this setup. Simply copying over the standard sof-byt-codec.tplg file does the trick, but then some mixer setting changes are necessary to fix the right speaker/headphones channel not working. Specifically I need to change the DACR channel setting from CH1 to CH0. But that mixer setting breaks the right channel when used with the sst-acpi driver. The problem seems to be that the codec and ucm-mixer settings expect DSP_B TDM 4 channel 24 bit format, but the standard sof-byt-codec.tplg file specifies I2S 2 channel 24 bit format. I tried changing the kernel to program the codec and SSP2 CPU DAIs both at I2S 2 channel 24 bit format to match what most other machine drivers are doing. This causes the no sound on right speaker channel to reproduce with the SST driver too. And like with the SOF driver it can be fixed by changing the mixer setting. With this kernel changes the SOF driver also still needs the mixer-setting fix. So I can get things to work, and make SOF/SST behave consistent by switching both DAIs to I2S 2 channel 24 bit format and then changing the mixer settings. The problem is that this will cause regressions unless users update their kernel and UCM in sync... So I tried to instead change the tplg file so that SOF will use DSP_B TDM 4 channel 24 bit format on the SSP2 DAI side, so that it matches what SST is doing and we don't need to change the mixer settings. This RFC series reflects my attempts at switching the SSP2 DAI to DSP_B TDM 4 channel 24 bit format when using SOF, unfortunately it does not work, the resulting tplg file causes the following errors: [ 21.537432] sof-audio-acpi 808622A8:00: Firmware info: version 1:6:0-18fab [ 21.537447] sof-audio-acpi 808622A8:00: Firmware: ABI 3:17:0 Kernel ABI 3:18:0 [ 25.003836] sof-audio-acpi 808622A8:00: Topology: ABI 3:17:0 Kernel ABI 3:18:0 [ 25.216003] sof-audio-acpi 808622A8:00: error: ipc error for 0x80010000 size 12 [ 25.216060] sof-audio-acpi 808622A8:00: error: failed to set DAI config for SSP2-Codec index 2 [ 25.216070] sof-audio-acpi 808622A8:00: error: failed to save DAI config for SSP2 [ 25.216080] sof-audio-acpi 808622A8:00: ASoC: physical link loading failed [ 25.216089] sof-audio-acpi 808622A8:00: ASoC: topology: could not load header: -22 [ 25.217459] sof-audio-acpi 808622A8:00: error: tplg component load failed -22 [ 25.217643] sof-audio-acpi 808622A8:00: error: failed to load DSP topology -22 Which I'm not getting when using the default sof-byt-codec.tplg file. Possible ways to fix this which I can think of are: 1. Fix SOF and/or the tplg file so that it can do DSP_B TDM 4 channel 24 bit on BYT/CHT. 2. Switch the machine driver over the I2S 2 channel 24 bit, to align it with other machine drivers; and update the UCM files to match. Preferably together with some string in the components string advertised by the kernel so that the UCM files can be written so that they also keep working with older kernels. 3. Keep using DSP_B TDM 4 channel 24 bit with sst-acpi and switch to I2S 2 channel 24 bit with sof-audio-acpi; and update the UCM file to set the mixer settings accordingly depending on which driver is in use. I'm not sure what the best way to proceed is. To go with solution 1 I'm going to need some help, since I'm stuck there. I find solution 2 tempting because I like the idea of standardising all BYT/CHT machine drivers on using I2S 2 channel 24 bit format between the SSP DAI and the codec DAI. But this means that people upgrading their kernel might see a regression until they update their UCM files too. Regards, Hans

2 4

Re: wm8960: add DAC Slope switch
by Charles Keepax 07 Dec '20

07 Dec '20

On Tue, Nov 24, 2020 at 06:23:13PM +0100, Lukas F. Hartmann wrote: > The WM8960 DAC has a "DAC Slope" switch that can toggle between two > different output filter curves. This patch adds support for it. > > Signed-off-by: Lukas F. Hartmann <lukas(a)mntre.com> > --- > +static int wm8960_put_dacslope(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); > + struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); > + unsigned int val = ucontrol->value.integer.value[0]; > + > + if (val > 1) > + return -EINVAL; > + > + wm8960->dacslope = val; > + > + return snd_soc_component_update_bits(component, WM8960_DACCTL1, > + 0x2, val<<1); Is the DAC Slope switch not in DACCTL2? Thanks, Charles

1 0

snd-rn-pci-acp3x - auto dmic detection unreliable
by Jaroslav Kysela 07 Dec '20

07 Dec '20

Hi, it seems that the ACPI auto-detection of the presence the digital microphone connected to AMD Renoir audio bridge is unreliable. It may be that hardware vendors do not provide the correct ACPI tables. Could you clarify the usage of _WOV integer value? It seems that this value is set on those plaforms which do not have the digital microphone (the microphone is connected to the HDA codec): Lenovo IdeaPad 5 (15ARE05) Lenovo Thinkpad X395 In any case, we should probably improve the auto-detection using DMI or other platform info. Jaroslav BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

3 6

[PATCH] ALSA: hda/realtek - Enable headset mic of ASUS X430UN with ALC256
by Chris Chiu 07 Dec '20

07 Dec '20

The ASUS laptop X430UN with ALC256 can't detect the headset microphone until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2398721ac1e..d5e4d0ba1008 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7827,6 +7827,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), -- 2.20.1

2 1

rt715: init: setup ADC07 to a proper volume
by GitHub pull_request - opened 07 Dec '20

07 Dec '20

alsa-project/alsa-ucm-conf pull request #65 was opened from libinyang: This patch initiates 'rt715 ADC 07 Capture Volume' to a proper volume. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/65 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/65.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: rawmidi: Access runtime->avail always in spinlock
by Takashi Iwai 06 Dec '20

06 Dec '20

The runtime->avail field may be accessed concurrently while some places refer to it without taking the runtime->lock spinlock, as detected by KCSAN. Usually this isn't a big problem, but for consistency and safety, we should take the spinlock at each place referencing this field. Reported-by: syzbot+a23a6f1215c84756577c(a)syzkaller.appspotmail.com Reported-by: syzbot+3d367d1df1d2b67f5c19(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/rawmidi.c | 49 +++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c78720a3299c..257ad5206240 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -95,11 +95,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -1019,7 +1029,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1098,7 +1108,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1115,9 +1125,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1255,7 +1267,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1444,9 +1456,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1526,6 +1540,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1544,13 +1559,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1568,13 +1586,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } -- 2.26.2

1 0

[PATCH] ALSA: seq: Use bool for snd_seq_queue internal flags
by Takashi Iwai 06 Dec '20

06 Dec '20

The snd_seq_queue struct contains various flags in the bit fields. Those are categorized to two different use cases, both of which are protected by different spinlocks. That implies that there are still potential risks of the bad operations for bit fields by concurrent accesses. For addressing the problem, this patch rearranges those flags to be a standard bool instead of a bit field. Reported-by: syzbot+63cbe31877bb80ef58f5(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 1c3a8d3254d9..c69105dc1a10 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -26,10 +26,10 @@ struct snd_seq_queue { struct snd_seq_timer *timer; /* time keeper for this queue */ int owner; /* client that 'owns' the timer */ - unsigned int locked:1, /* timer is only accesibble by owner if set */ - klocked:1, /* kernel lock (after START) */ - check_again:1, - check_blocked:1; + bool locked; /* timer is only accesibble by owner if set */ + bool klocked; /* kernel lock (after START) */ + bool check_again; /* concurrent access happened during check */ + bool check_blocked; /* queue being checked */ unsigned int flags; /* status flags */ unsigned int info_flags; /* info for sync */ -- 2.26.2

1 0

[PATCH 0/1] dt-bindings: eliminate yamllint warnings
by Zhen Lei 06 Dec '20

06 Dec '20

There're too many people, I just send to the maintainer, reviewer, supporter. Eliminate below warnings: ./Documentation/devicetree/bindings/clock/imx8qxp-lpcg.yaml:32:13: [warning] wrong indentation: expected 14 but found 12 (indentation) ./Documentation/devicetree/bindings/clock/imx8qxp-lpcg.yaml:35:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/display/intel,keembay-msscam.yaml:21:6: [warning] wrong indentation: expected 6 but found 5 (indentation) ./Documentation/devicetree/bindings/display/bridge/analogix,anx7625.yaml:52:9: [warning] wrong indentation: expected 6 but found 8 (indentation) ./Documentation/devicetree/bindings/display/bridge/intel,keembay-dsi.yaml:42:8: [warning] wrong indentation: expected 8 but found 7 (indentation) ./Documentation/devicetree/bindings/display/bridge/intel,keembay-dsi.yaml:45:8: [warning] wrong indentation: expected 8 but found 7 (indentation) ./Documentation/devicetree/bindings/display/panel/novatek,nt36672a.yaml:25:10: [warning] wrong indentation: expected 10 but found 9 (indentation) ./Documentation/devicetree/bindings/media/i2c/adv7604.yaml:24:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:4:1: [error] missing document start "---" (document-start) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:29:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:32:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/ovti,ov772x.yaml:79:17: [warning] wrong indentation: expected 14 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/ovti,ov772x.yaml:88:17: [warning] wrong indentation: expected 14 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/sony,imx214.yaml:72:17: [warning] wrong indentation: expected 18 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/sony,imx214.yaml:75:17: [warning] wrong indentation: expected 18 but found 16 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:20:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:30:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:33:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml:10:4: [warning] wrong indentation: expected 2 but found 3 (indentation) Zhen Lei (1): dt-bindings: eliminate yamllint warnings .../devicetree/bindings/clock/imx8qxp-lpcg.yaml | 20 ++++++++--------- .../bindings/display/bridge/analogix,anx7625.yaml | 4 ++-- .../bindings/display/bridge/intel,keembay-dsi.yaml | 4 ++-- .../bindings/display/intel,keembay-msscam.yaml | 4 ++-- .../bindings/display/panel/novatek,nt36672a.yaml | 2 +- .../devicetree/bindings/media/i2c/adv7604.yaml | 4 ++-- .../devicetree/bindings/media/i2c/mipi-ccs.yaml | 11 ++++----- .../devicetree/bindings/media/i2c/ovti,ov772x.yaml | 12 +++++----- .../devicetree/bindings/media/i2c/sony,imx214.yaml | 12 +++++----- Documentation/devicetree/bindings/mmc/mtk-sd.yaml | 26 +++++++++++----------- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++-- 11 files changed, 52 insertions(+), 51 deletions(-) -- 1.8.3

4 5

thinkpad t14s amd: unable to control built-in mic and headphones mic independently
by GitHub issues - opened 05 Dec '20

05 Dec '20

alsa-project/alsa-ucm-conf issue #64 was opened from tanuk: I did some investigation on a [pulseaudio bug report](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1067), and found that the UCM configuration for the internal mic and headset mic both use "Capture" as the CaptureMixerElem. This means that muting one device mutes also the other (and I guess volume control is shared too). In amixer output I don't see a separate control for the headset mic, but for the internal mic volume I guess "Digital" can be used (there's no switch, though). So it seems that these changes are needed: - don't set CaptureMixerElem, CaptureSwitch or CaptureVolume for the headset mic - set the internal mic CaptureMixerElem and CaptureVolume to Digital - you may want to also set CaptureMasterElem for both both mics to Capture Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/64 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Takashi Iwai 05 Dec '20

05 Dec '20

On Thu, 03 Dec 2020 21:06:24 +0100, Ben Bell wrote: > > On Sat, Nov 28, 2020 at 09:36:00AM +0000, Ben Bell wrote: > > > In general you should avoid 44.1kHz if you want a small period size > > > for a realtime process on USB-audio. With 44.1kHz, the packet size > > > can't be fixed in integer, and the ISO transfer requires variable > > > packet sizes. OTOH, ALSA API requires the fixed period size, hence > > > it'll lead to inconsistencies occasionally. > > So changing to 48kHz had no appreciable effect, but I'm working at that > rate for now to eliminate the 44.1kHz weirdness from investigations. > Using a USB2 card, also had no effect. Testing with a different USB audio > interface (albeit a simple stereo one) didn't exhibit the behaviour, even > when I took buffer sizes right down. > > I'm not sure where to go to get more information or where's the best place > to ask for help. I'm happy to do the leg work, but I don't know enough > about the kernel, alsa or USB to figure it out without some help. > > Current question: what is the delay in /proc/asound/card1/pcm0c/sub0/status > actually measuring? I'm assuming it's measured in samples? I've written > something to scrape the stats out in a tight loop and report. > > What I see is a cycle where the delay rises and then a chunk equal to the > frames per period (or sometimes, earlier on fpp-48) is removed. That feels > like chunks being read out of a buffer. All fine. > > After a while though, the maximum delay we reach with each cycle is creeping > up. It increases by one every few cycles (usually two or three, or three or > four -- always oscillating between two values) but of it's still only being > emptied by a full period's worth of samples each cycle. So the overall effect > is the delay creeps up and up until it hits the buffer size and then we get > an xrun. > > Like I said in the initial email, it feels like some sort of clock drift > problem, where we're managing very slowly to collect more samples than > we're reading -- to the tune of about 1 extra every few cycles -- and > nothing on the consumer side is ever managing to compensate for that. > I'm not even sure how that sort of drift would be possible though. Seems > surprising. > > Does any of this sound suspicious, or for that matter completely normal? > Any suggestions where should I be looking next? At least you can try the latest patch set destined for 5.11, which should improve the cases for the implicit feedback. The patches are either in linux-next tree or the topic/usb-audio-refactoring branch of my sound.git tree. It's based on 5.10-rc, so should be cleanly mergeable to the latest Linus tree. Takashi

2 1

[PATCH] slimbus: qcom-ngd-ctrl: Fix error handling to keep reference balanced
by Zhang Qilong 05 Dec '20

05 Dec '20

In the error branch, we should call pm_runtime_put_noidle and pm_runtime_disable to decrease usage counter and increase the power disable depth to avoid reference leaks. Fixes: 917809e2280bb ("slimbus: ngd: Add qcom SLIMBus NGD driver") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..a9b4a1d95045 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1483,7 +1483,7 @@ static int qcom_slim_ngd_probe(struct platform_device *pdev) ret = qcom_slim_ngd_qmi_svc_event_init(ctrl); if (ret) { dev_err(&pdev->dev, "QMI service registration failed:%d", ret); - return ret; + goto pm_err; } INIT_WORK(&ctrl->m_work, qcom_slim_ngd_master_worker); @@ -1500,6 +1500,9 @@ static int qcom_slim_ngd_probe(struct platform_device *pdev) qcom_slim_ngd_qmi_svc_event_deinit(&ctrl->qmi); if (ctrl->mwq) destroy_workqueue(ctrl->mwq); +pm_err: + pm_runtime_put_noidle(dev); + pm_runtime_disable(dev); return ret; } -- 2.25.4

1 0

[PATCH] soundwire: intel: fix another unused-function warning
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> Without CONFIG_PM, there is another warning about an unused function: drivers/soundwire/intel.c:530:12: error: 'intel_link_power_down' defined but not used [-Werror=unused-function] After a previous fix, the driver already uses both an #ifdef and a __maybe_unused annotation but still gets it wrong. Remove the ifdef and instead use __maybe_unused consistently to avoid the problem for good. Fixes: f046b2334083 ("soundwire: intel: fix intel_suspend/resume defined but not used warning") Fixes: ebf878eddbb4 ("soundwire: intel: add pm_runtime support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/soundwire/intel.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 6a1e862b16c3..66adb258a425 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1585,8 +1585,6 @@ int intel_master_process_wakeen_event(struct platform_device *pdev) * PM calls */ -#ifdef CONFIG_PM - static int __maybe_unused intel_suspend(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); @@ -1641,7 +1639,7 @@ static int __maybe_unused intel_suspend(struct device *dev) return 0; } -static int intel_suspend_runtime(struct device *dev) +static int __maybe_unused intel_suspend_runtime(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); @@ -1796,7 +1794,7 @@ static int __maybe_unused intel_resume(struct device *dev) return ret; } -static int intel_resume_runtime(struct device *dev) +static int __maybe_unused intel_resume_runtime(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); @@ -1969,8 +1967,6 @@ static int intel_resume_runtime(struct device *dev) return ret; } -#endif - static const struct dev_pm_ops intel_pm = { SET_SYSTEM_SLEEP_PM_OPS(intel_suspend, intel_resume) SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) -- 2.27.0

3 2

[PATCH v2] ALSA: hda/realtek: make bass spk volume adjustable on a yoga laptop
by Hui Wang 05 Dec '20

05 Dec '20

This change could fix 2 issues on this machine: - the bass speaker's output volume can't be adjusted, that is because the bass speaker is routed to the DAC (Nid 0x6) which has no volume control. - after plugging a headset with vol+, vol- and pause buttons on it, press those buttons, nothing happens, this means those buttons don't work at all. This machine has alc287 codec, need to add the codec id to the disable/enable_headset_jack_key(), then the headset button could work. The quirk of ALC285_FIXUP_THINKPAD_HEADSET_JACK could fix both of these 2 issues. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..5a905fa1b33a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3104,6 +3104,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3130,6 +3131,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -8578,6 +8580,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK, + {0x14, 0x90170110}, + {0x17, 0x90170111}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: make bass spk volume adjustable on a yoga laptop
by Hui Wang 05 Dec '20

05 Dec '20

This also make the headset button work on this machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..5a905fa1b33a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3104,6 +3104,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3130,6 +3131,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -8578,6 +8580,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK, + {0x14, 0x90170110}, + {0x17, 0x90170111}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.25.1

2 2

[PATCH] ASoC: jz4740-i2s: add missed checks for clk_get()
by Chuhong Yuan 05 Dec '20

05 Dec '20

jz4740_i2s_set_sysclk() does not check the return values of clk_get(), while the file dereferences the pointers in clk_put(). Add the missed checks to fix it. Fixes: 11bd3dd1b7c2 ("ASoC: Add JZ4740 ASoC support") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index c7bd20104b20..0793e284d0e7 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -312,10 +312,14 @@ static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, switch (clk_id) { case JZ4740_I2S_CLKSRC_EXT: parent = clk_get(NULL, "ext"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); break; case JZ4740_I2S_CLKSRC_PLL: parent = clk_get(NULL, "pll half"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); ret = clk_set_rate(i2s->clk_i2s, freq); break; -- 2.26.2

3 2

[PATCH v2] ASoC: intel: sof_rt5682: Add support for tgl_rt1011_rt5682
by Brent Lu 05 Dec '20

05 Dec '20

This patch adds the driver data for two rt1011 speaker amplifiers on SSP1 and rt5682 on SSP0 for TGL platform. DAI format for rt1011 is leveraged from cml_rt1011_rt5682 which is 4-slot tdm with 100fs bclk. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 138 ++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 24 +++ sound/soc/intel/boards/sof_rt5682.c | 26 +++- .../intel/common/soc-acpi-intel-tgl-match.c | 17 ++- 6 files changed, 201 insertions(+), 7 deletions(-) create mode 100644 sound/soc/intel/boards/sof_realtek_common.c create mode 100644 sound/soc/intel/boards/sof_realtek_common.h diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index dddb672a6d55..b58b9b60d37e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -443,6 +443,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT5682_I2C select SND_SOC_DMIC diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a58e4d22e9c8..5f03e484b215 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -18,7 +18,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o +snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c new file mode 100644 index 000000000000..f3cf73c620ba --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -0,0 +1,138 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2020 Intel Corporation. All rights reserved. + +#include <linux/device.h> +#include <linux/kernel.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <uapi/sound/asound.h> +#include "../../codecs/rt1011.h" +#include "sof_realtek_common.h" + +/* + * Current only 2-amp configuration is supported for rt1011 + */ +static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left SPO" }, + { "Right Spk", NULL, "Right SPO" }, +}; + +/* + * Make sure device's Unique ID follows this configuration: + * + * Two speakers: + * 0: left, 1: right + * Four speakers: + * 0: Woofer left, 1: Woofer right + * 2: Tweeter left, 3: Tweeter right + */ +static struct snd_soc_codec_conf rt1011_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1011_dai_link_components[] = { + { + .name = RT1011_DEV0_NAME, + .dai_name = RT1011_CODEC_DAI, + }, + { + .name = RT1011_DEV1_NAME, + .dai_name = RT1011_CODEC_DAI, + }, +}; + +static const struct { + unsigned int tx; + unsigned int rx; +} rt1011_tdm_mask[] = { + {.tx = 0x4, .rx = 0x1}, + {.tx = 0x8, .rx = 0x2}, +}; + +static int rt1011_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int srate, i, ret = 0; + + srate = params_rate(params); + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + /* 100 Fs to drive 24 bit data */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, + 100 * srate, 256 * srate); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set pll, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1011_FS_SYS_PRE_S_PLL1, + 256 * srate, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set sysclk, ret %d\n", + ret); + return ret; + } + + if (i >= ARRAY_SIZE(rt1011_tdm_mask)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", + i); + return -ENODEV; + } + + ret = snd_soc_dai_set_tdm_slot(codec_dai, rt1011_tdm_mask[i].tx, + rt1011_tdm_mask[i].rx, 4, + params_width(params)); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", + ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_ops rt1011_ops = { + .hw_params = rt1011_hw_params, +}; + +static int rt1011_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_dapm_routes, + ARRAY_SIZE(rt1011_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1011_dai_link_components; + link->num_codecs = ARRAY_SIZE(rt1011_dai_link_components); + link->init = rt1011_init; + link->ops = &rt1011_ops; +} + +void sof_rt1011_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1011_codec_confs; + card->num_configs = ARRAY_SIZE(rt1011_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h new file mode 100644 index 000000000000..87cb3812b926 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright(c) 2020 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with Realtek Codecs. + */ +#ifndef __SOF_REALTEK_COMMON_H +#define __SOF_REALTEK_COMMON_H + +#include <sound/soc.h> + +#define RT1011_CODEC_DAI "rt1011-aif" +#define RT1011_DEV0_NAME "i2c-10EC1011:00" +#define RT1011_DEV1_NAME "i2c-10EC1011:01" +#define RT1011_DEV2_NAME "i2c-10EC1011:02" +#define RT1011_DEV3_NAME "i2c-10EC1011:03" + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link); +void sof_rt1011_codec_conf(struct snd_soc_card *card); + +#endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 891f908659f5..8b1ca2da9bb9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -24,6 +24,7 @@ #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" #include "sof_maxim_common.h" +#include "sof_realtek_common.h" #define NAME_SIZE 32 @@ -41,10 +42,11 @@ #define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) -#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) -#define SOF_RT1015_SPEAKER_AMP_100FS BIT(14) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) +#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -741,6 +743,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].codecs = max98360a_component; links[id].num_codecs = ARRAY_SIZE(max98360a_component); links[id].init = speaker_codec_init; + } else if (sof_rt5682_quirk & + SOF_RT1011_SPEAKER_AMP_PRESENT) { + sof_rt1011_dai_link(&links[id]); } else { links[id].codecs = max98357a_component; links[id].num_codecs = ARRAY_SIZE(max98357a_component); @@ -851,6 +856,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) sof_max98373_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) + sof_rt1011_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -931,6 +938,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT1015_SPEAKER_AMP_100FS | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "tgl_rt1011_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1011_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; @@ -948,6 +964,7 @@ module_platform_driver(sof_audio) MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); @@ -955,3 +972,4 @@ MODULE_ALIAS("platform:jsl_rt5682_rt1015"); MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); +MODULE_ALIAS("platform:tgl_rt1011_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 9f243e60b95c..98196e9fad62 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -9,7 +9,7 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> -static struct snd_soc_acpi_codecs tgl_codecs = { +static const struct snd_soc_acpi_codecs tgl_codecs = { .num_codecs = 1, .codecs = {"MX98357A"} }; @@ -305,11 +305,16 @@ static const struct snd_soc_acpi_link_adr tgl_3_in_1_sdca[] = { {} }; -static struct snd_soc_acpi_codecs tgl_max98373_amp = { +static const struct snd_soc_acpi_codecs tgl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} }; +static const struct snd_soc_acpi_codecs tgl_rt1011_amp = { + .num_codecs = 1, + .codecs = {"10EC1011"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", @@ -335,6 +340,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "tgl_rt1011_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_rt1011_amp, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); -- 2.17.1

3 2

[PATCH] ASoC: fsl_aud2htx: mark PM functions as __maybe_unused
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> When CONFIG_PM is disabled, we get a warning for unused functions: sound/soc/fsl/fsl_aud2htx.c:261:12: error: unused function 'fsl_aud2htx_runtime_suspend' [-Werror,-Wunused-function] static int fsl_aud2htx_runtime_suspend(struct device *dev) sound/soc/fsl/fsl_aud2htx.c:271:12: error: unused function 'fsl_aud2htx_runtime_resume' [-Werror,-Wunused-function] static int fsl_aud2htx_runtime_resume(struct device *dev) Mark these as __maybe_unused to avoid the warning without adding an #ifdef. Fixes: 8a24c834c053 ("ASoC: fsl_aud2htx: Add aud2htx module driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/fsl_aud2htx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 4091ccc7c3e9..d70d5e75a30c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -258,7 +258,7 @@ static int fsl_aud2htx_remove(struct platform_device *pdev) return 0; } -static int fsl_aud2htx_runtime_suspend(struct device *dev) +static int __maybe_unused fsl_aud2htx_runtime_suspend(struct device *dev) { struct fsl_aud2htx *aud2htx = dev_get_drvdata(dev); @@ -268,7 +268,7 @@ static int fsl_aud2htx_runtime_suspend(struct device *dev) return 0; } -static int fsl_aud2htx_runtime_resume(struct device *dev) +static int __maybe_unused fsl_aud2htx_runtime_resume(struct device *dev) { struct fsl_aud2htx *aud2htx = dev_get_drvdata(dev); int ret; -- 2.27.0

3 2

[PATCH] ASoC: atmel: mchp-spdifrx needs COMMON_CLK
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> Compile-testing this driver on an older platform without CONFIG_COMMON_CLK fails with ERROR: modpost: "clk_set_min_rate" [sound/soc/atmel/snd-soc-mchp-spdifrx.ko] undefined! Make this is a strict dependency. Fixes: ef265c55c1ac ("ASoC: mchp-spdifrx: add driver for SPDIF RX") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index bd8854bfd2ee..142373ec411a 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -148,6 +148,7 @@ config SND_MCHP_SOC_SPDIFTX config SND_MCHP_SOC_SPDIFRX tristate "Microchip ASoC driver for boards using S/PDIF RX" depends on OF && (ARCH_AT91 || COMPILE_TEST) + depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- 2.27.0

3 2

[PATCH] ASoC: cros_ec_codec: fix uninitialized memory read
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> gcc points out a memory area that is copied to a device but not initialized: sound/soc/codecs/cros_ec_codec.c: In function 'i2s_rx_event': arch/x86/include/asm/string_32.h:83:20: error: '*((void *)&p+4)' may be used uninitialized in this function [-Werror=maybe-uninitialized] 83 | *((int *)to + 1) = *((int *)from + 1); Initialize all the unused fields to zero. Fixes: 727f1c71c780 ("ASoC: cros_ec_codec: refactor I2S RX") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cros_ec_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 58894bf47514..f33a2a9654e7 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -332,7 +332,7 @@ static int i2s_rx_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_to_component(w->dapm); struct cros_ec_codec_priv *priv = snd_soc_component_get_drvdata(component); - struct ec_param_ec_codec_i2s_rx p; + struct ec_param_ec_codec_i2s_rx p = {}; switch (event) { case SND_SOC_DAPM_PRE_PMU: -- 2.27.0

3 4

[PATCH] ASoC: qcom: fix QDSP6 dependencies, attempt #3
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> The previous fix left another warning in randconfig builds: WARNING: unmet direct dependencies detected for SND_SOC_QDSP6 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && QCOM_APR [=y] && COMMON_CLK [=n] Selected by [y]: - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && QCOM_APR [=y] Add one more dependency for this one. Fixes: 2bc8831b135c ("ASoC: qcom: fix SDM845 & QDSP6 dependencies more") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 27f93006be96..cc7c1de2f1d9 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -106,6 +106,7 @@ config SND_SOC_QDSP6 config SND_SOC_MSM8996 tristate "SoC Machine driver for MSM8996 and APQ8096 boards" depends on QCOM_APR + depends on COMMON_CLK select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help -- 2.27.0

3 2

[PATCH] ASoC: q6afe-clocks: Add missing parent clock rate
by Srinivas Kandagatla 05 Dec '20

05 Dec '20

setting clock rate on child clocks without a parent clock rate will result in zero clk rate for child. This also means that when audio is started dsp will attempt to access registers without enabling clock resulting in board boot up. Fix this by adding the missing parent clock rate. Fixes: 520a1c396d196 ("ASoC: q6afe-clocks: add q6afe clock controller") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-clocks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-clocks.c b/sound/soc/qcom/qdsp6/q6afe-clocks.c index 87e4633afe2c..f0362f061652 100644 --- a/sound/soc/qcom/qdsp6/q6afe-clocks.c +++ b/sound/soc/qcom/qdsp6/q6afe-clocks.c @@ -16,6 +16,7 @@ .afe_clk_id = Q6AFE_##id, \ .name = #id, \ .attributes = LPASS_CLK_ATTRIBUTE_COUPLE_NO, \ + .rate = 19200000, \ .hw.init = &(struct clk_init_data) { \ .ops = &clk_q6afe_ops, \ .name = #id, \ -- 2.21.0

2 1

[PATCH] ASoC: SOF: control: fix cppcheck warning in snd_sof_volume_info()
by Kai Vehmanen 05 Dec '20

05 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Fix cppcheck warning: sound/soc/sof/control.c:117:82: style:inconclusive: Function 'snd_sof_volume_info' argument 2 names different: declaration 'ucontrol' definition 'uinfo'. [funcArgNamesDifferent] Fixes: fca18e62984a ("ASoC: SOF: control: override volume info callback") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/sof-audio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index d00a918b6f39..dc930fc2f4b5 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -125,7 +125,7 @@ int snd_sof_volume_get(struct snd_kcontrol *kcontrol, int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *ucontrol); + struct snd_ctl_elem_info *uinfo); int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_switch_put(struct snd_kcontrol *kcontrol, -- 2.28.0

2 1

Re: [PATCH 1/4] ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver
by Mark Brown 04 Dec '20

04 Dec '20

On Fri, Dec 04, 2020 at 02:42:54AM +0000, Shuming [范書銘] wrote: > > > +/* RT1316 SDCA function topology */ > > > +#define FUN_SMART_AMP 0x04 > > Full marks for picking this constant! > Sorry, I don't get that point. Could you give me a hint? The name is funny - "fun smart amp".

2 1

[PATCH] [RFC] alsaloop: add feedback frequency control support for UAC2 gadgets
by Ruslan Bilovol 04 Dec '20

04 Dec '20

Add support of special "PCM Feedback Frequency Hz" UAC2 Gadget mixer control that is designed to notify host about real sampling frequency of the gadget so it can adjust number of samples that hosts sends to the gadget. This is useful if both host and gadget has its own internal freerunning clock, so host can adjust number of samples sent, preventing overrun/underrun conditions. This patch reuses logic of the "PCM Rate Shift 100000" control used in case of in-kernel ALSA loopback driver. The only difference is alsaloop reports not rate shift but frequency in Hz Signed-off-by: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> --- alsaloop/alsaloop.h | 1 + alsaloop/pcmjob.c | 35 +++++++++++++++++++++++++---------- 2 files changed, 26 insertions(+), 10 deletions(-) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index c4aa618..9a50a42 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -122,6 +122,7 @@ struct loopback_handle { unsigned int ctl_pollfd_count; snd_ctl_elem_value_t *ctl_notify; snd_ctl_elem_value_t *ctl_rate_shift; + snd_ctl_elem_value_t *ctl_fback_freq; snd_ctl_elem_value_t *ctl_active; snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 6a9aff4..b3802a8 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1058,15 +1058,22 @@ static int set_notify(struct loopback_handle *lhandle, int enable) static int set_rate_shift(struct loopback_handle *lhandle, double pitch) { - int err; + int err = 0; - if (lhandle->ctl_rate_shift == NULL) - return 0; - snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, pitch * 100000); - err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_rate_shift); - if (err < 0) { - logit(LOG_CRIT, "Cannot set PCM Rate Shift element for %s: %s\n", lhandle->id, snd_strerror(err)); - return err; + if (lhandle->ctl_rate_shift) { + snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, pitch * 100000); + err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_rate_shift); + if (err < 0) { + logit(LOG_CRIT, "Cannot set PCM Rate Shift element for %s: %s\n", lhandle->id, snd_strerror(err)); + return err; + } + } else if (lhandle->ctl_fback_freq) { + snd_ctl_elem_value_set_integer(lhandle->ctl_fback_freq, 0, lhandle->rate * (2.0 - pitch)); + err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_fback_freq); + if (err < 0) { + logit(LOG_CRIT, "Cannot set PCM Feedback Frequency element for %s: %s\n", lhandle->id, snd_strerror(err)); + return err; + } } return 0; } @@ -1195,6 +1202,7 @@ static int openctl(struct loopback_handle *lhandle, int device, int subdevice) int err; lhandle->ctl_rate_shift = NULL; + lhandle->ctl_fback_freq = NULL; if (lhandle->loopback->play == lhandle) { if (lhandle->loopback->controls) goto __events; @@ -1204,6 +1212,8 @@ static int openctl(struct loopback_handle *lhandle, int device, int subdevice) &lhandle->ctl_notify); openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", &lhandle->ctl_rate_shift); + openctl_elem(lhandle, device, subdevice, "PCM Feedback Frequency Hz", + &lhandle->ctl_fback_freq); set_rate_shift(lhandle, 1); openctl_elem(lhandle, device, subdevice, "PCM Slave Active", &lhandle->ctl_active); @@ -1289,6 +1299,9 @@ static int closeit(struct loopback_handle *lhandle) if (lhandle->ctl_rate_shift) snd_ctl_elem_value_free(lhandle->ctl_rate_shift); lhandle->ctl_rate_shift = NULL; + if (lhandle->ctl_fback_freq) + snd_ctl_elem_value_free(lhandle->ctl_fback_freq); + lhandle->ctl_fback_freq = NULL; if (lhandle->ctl) err = snd_ctl_close(lhandle->ctl); lhandle->ctl = NULL; @@ -1334,9 +1347,11 @@ int pcmjob_init(struct loopback *loop) snprintf(id, sizeof(id), "%s/%s", loop->play->id, loop->capt->id); id[sizeof(id)-1] = '\0'; loop->id = strdup(id); - if (loop->sync == SYNC_TYPE_AUTO && loop->capt->ctl_rate_shift) + if (loop->sync == SYNC_TYPE_AUTO && (loop->capt->ctl_rate_shift || + loop->capt->ctl_fback_freq)) loop->sync = SYNC_TYPE_CAPTRATESHIFT; - if (loop->sync == SYNC_TYPE_AUTO && loop->play->ctl_rate_shift) + if (loop->sync == SYNC_TYPE_AUTO && (loop->play->ctl_rate_shift || + loop->play->ctl_fback_freq)) loop->sync = SYNC_TYPE_PLAYRATESHIFT; #ifdef USE_SAMPLERATE if (loop->sync == SYNC_TYPE_AUTO && loop->src_enable) -- 1.9.1

2 2

How to set channel balance in amixer?
by GitHub issues - opened 04 Dec '20

04 Dec '20

alsa-project/alsa-utils issue #73 was opened from quantum-booty: When I change volumn using keyboard media keys, the left and right channel becomes imbalanced. ![image](https://user-images.githubusercontent.com/28958399/101188833-23befc00-364e-11eb-8816-efd94b9514d4.png) I currently have this mapping in i3, I would like to add command to balance the channels after I change the volumn. bindsym XF86AudioRaiseVolume exec --no-startup-id "amixer set Master '5%+,5%+'" Issue URL : https://github.com/alsa-project/alsa-utils/issues/73 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] [v2] ASoC: cros_ec_codec: fix uninitialized memory read
by Arnd Bergmann 04 Dec '20

04 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> gcc points out a memory area that is copied to a device but not initialized: sound/soc/codecs/cros_ec_codec.c: In function 'i2s_rx_event': arch/x86/include/asm/string_32.h:83:20: error: '*((void *)&p+4)' may be used uninitialized in this function [-Werror=maybe-uninitialized] 83 | *((int *)to + 1) = *((int *)from + 1); Change the length of the command to only pass down the part of the structure that has been initialized, as Tzung-Bi Shih explains that only that member is meant to be used. Cc: Tzung-Bi Shih <tzungbi(a)google.com> Fixes: 727f1c71c780 ("ASoC: cros_ec_codec: refactor I2S RX") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cros_ec_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 58894bf47514..6ec673573c70 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -348,7 +348,7 @@ static int i2s_rx_event(struct snd_soc_dapm_widget *w, } return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, - (uint8_t *)&p, sizeof(p), NULL, 0); + &p.cmd, sizeof(p.cmd), NULL, 0); } static struct snd_soc_dapm_widget i2s_rx_dapm_widgets[] = { -- 2.27.0

2 2

Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Ben Bell 03 Dec '20

03 Dec '20

I've got a Behringer WING digital mixer which is equipped with a USB interface supporting 48 in, 48 out at 44.1 or 48kHz. It's plugged into a USB3 interface and for the most part it seems to work well as a class-compliant audio interface, but I'm struggling to eliminate xruns and it's starting to feel either driver or hardware quirk related issue. I've done all the usual things -- I'm running a PREEMPT_RT kernel, set up the realtime priorities of Jack and the relevant USB IRQ to no avail. I'm doing most of the debugging remotely over ssh with X shutdown and most other processes on the system stopped. There's nothing plugged into the computer beyond network, keyboard, mouse and the audio interface. I'm currently testing on kernel 5.10.0-rc5 with preempt-rt patches, but I've previously seen the same things on 5.4-rt (Debian backports) and 5.8 (self-compiled). The xruns are characterised by bursts of "retire_capture_urb" warnings in the kernel logs (and in particular the frame status for everything in those URBs is EXDEV, meaning I think that the frames haven't been consumed by the time it's retired). I've patched sound/driver/pcm.c to provide me with a bit more debugging information and it looks like usually it's almost always between 7-10 consecutive calls to retire_capture_urb that are affected. The bursts of retire_capture_urb warnings seem roughly cyclic, with a cycle time that is dependent on the combination of the frames/period setting and the number of periods per buffer, though it doesn't appear to be a strictly linear thing: 512/2 (~170s cycle); 256/2 (~94s); 128/3 (~65s); 64/13 (~270s). I can't immediately see any definitive smoking gun in the urb or interrupt counts, but there is some grouping in the timings between the xruns. Taking 64 frames, 5 periods, it's usually 50s, but sometimes 60s: 21:47:57 50s since xrun 4 21:48:47 50s since xrun 5 21:49:46 59s since xrun 6 21:50:46 60s since xrun 7 21:51:36 50s since xrun 8 21:52:36 60s since xrun 9 21:53:36 60s since xrun 10 21:54:37 61s since xrun 11 21:55:27 50s since xrun 12 21:56:27 60s since xrun 13 21:57:27 60s since xrun 14 21:58:17 50s since xrun 15 21:59:16 59s since xrun 16 22:00:06 50s since xrun 17 22:00:56 50s since xrun 18 ... If I count the URB numbers (sorry, I'm shaky on the terminology, but I'm counting each URB which is retired) I often (but not always) see the same number of URBs passing between xrun bursts. Currently at 64 frames/5 periods, it's often exactly 50042 or 49923 urbs between bursts. The delay between starting Jack and encountering the first burst is not predictable -- it's not just a whole cycle -- but stopping jack for e.g 30s then starting again delays the next burst by the same amount. So it does seem related to something in the audio streaming rather than anything else going on on the system interfering. It's as if something is slowly slipping out of sync between the Wing and the kernel until they need to resync, but I don't really know enough about USB to have any deeper insight. All the tests above happen to be at 44.1kHz with the Wing set to 48 in, 48 out, but changing to 48kHz or 2/2 IO doesn't seem to cure anything. One more data point which may not be relevant: if I switch off or unplug the Wing while Jack is running, the system gradually locks up over a period of a few seconds. I can get something out of the console briefly afterwards but within about ten seconds it's completely unresponsive. I imagine that pulling out a USB device from a realtime thread isn't a kind thing to do, but I don't recall this happening with other interfaces. Any help or insights into where I should be looking (I'm a newbie in kernel space) appreciated. If someone wants remote access to the box to investigate in realtime we could probably figure something out. bjb

2 3

[bug report] ASoC: mchp-spdifrx: add driver for SPDIF RX
by Dan Carpenter 03 Dec '20

03 Dec '20

Hello Codrin Ciubotariu, The patch ef265c55c1ac: "ASoC: mchp-spdifrx: add driver for SPDIF RX" from Oct 2, 2020, leads to the following static checker warning: sound/soc/atmel/mchp-spdifrx.c:468 mchp_spdifrx_hw_params() warn: 'dev->gclk' not released on lines: 468. sound/soc/atmel/mchp-spdifrx.c 442 params_format(params)); 443 return -EINVAL; 444 } 445 446 if (dev->gclk_enabled) { 447 clk_disable_unprepare(dev->gclk); 448 dev->gclk_enabled = 0; 449 } 450 ret = clk_set_min_rate(dev->gclk, params_rate(params) * 451 SPDIFRX_GCLK_RATIO_MIN + 1); 452 if (ret) { 453 dev_err(dev->dev, 454 "unable to set gclk min rate: rate %u * ratio %u + 1\n", 455 params_rate(params), SPDIFRX_GCLK_RATIO_MIN); 456 return ret; 457 } 458 ret = clk_prepare_enable(dev->gclk); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 459 if (ret) { 460 dev_err(dev->dev, "unable to enable gclk: %d\n", ret); 461 return ret; 462 } 463 dev->gclk_enabled = 1; 464 465 dev_dbg(dev->dev, "GCLK range min set to %d\n", 466 params_rate(params) * SPDIFRX_GCLK_RATIO_MIN + 1); 467 468 return regmap_write(dev->regmap, SPDIFRX_MR, mr); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Smatch is complaining that if the regmap_write() fails then we should disable and unprepare the "dev->gclk". 469 } regards, dan carpenter

2 2

[PATCH] ASoC: intel: sof_rt5682: Add support for tgl_rt1011_rt5682
by Brent Lu 03 Dec '20

03 Dec '20

This patch adds the driver data for two rt1011 speaker amplifiers on SSP1 and rt5682 on SSP0 for TGL platform. DAI format for rt1011 is leveraged from cml_rt1011_rt5682 which is 4-slot tdm with 100fs bclk. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 137 ++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 24 +++ sound/soc/intel/boards/sof_rt5682.c | 26 +++- .../intel/common/soc-acpi-intel-tgl-match.c | 13 ++ 6 files changed, 198 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/boards/sof_realtek_common.c create mode 100644 sound/soc/intel/boards/sof_realtek_common.h diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index dddb672a6d55..b58b9b60d37e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -443,6 +443,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT5682_I2C select SND_SOC_DMIC diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a58e4d22e9c8..5f03e484b215 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -18,7 +18,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o +snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c new file mode 100644 index 000000000000..b3bc41385436 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -0,0 +1,137 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2020 Intel Corporation. All rights reserved. + +#include <linux/kernel.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <uapi/sound/asound.h> +#include "../../codecs/rt1011.h" +#include "sof_realtek_common.h" + +/* + * Current only 2-amp configuration is supported for rt1011 + */ +static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left SPO" }, + { "Right Spk", NULL, "Right SPO" }, +}; + +/* + * Make sure device's Unique ID follows this configuration: + * + * Two speakers: + * 0: left, 1: right + * Four speakers: + * 0: Woofer left, 1: Woofer right + * 2: Tweeter left, 3: Tweeter right + */ +static struct snd_soc_codec_conf rt1011_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1011_dai_link_components[] = { + { + .name = RT1011_DEV0_NAME, + .dai_name = RT1011_CODEC_DAI, + }, + { + .name = RT1011_DEV1_NAME, + .dai_name = RT1011_CODEC_DAI, + }, +}; + +struct { + unsigned int tx; + unsigned int rx; +} rt1011_tdm_mask[] = { + {.tx = 0x4, .rx = 0x1}, + {.tx = 0x8, .rx = 0x2}, +}; + +static int rt1011_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int srate, i, ret = 0; + + srate = params_rate(params); + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + /* 100 Fs to drive 24 bit data */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, + 100 * srate, 256 * srate); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set pll, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1011_FS_SYS_PRE_S_PLL1, + 256 * srate, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set sysclk, ret %d\n", + ret); + return ret; + } + + if (i >= ARRAY_SIZE(rt1011_tdm_mask)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", + i); + return -ENODEV; + } + + ret = snd_soc_dai_set_tdm_slot(codec_dai, rt1011_tdm_mask[i].tx, + rt1011_tdm_mask[i].rx, 4, + params_width(params)); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", + ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_ops rt1011_ops = { + .hw_params = rt1011_hw_params, +}; + +static int rt1011_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_dapm_routes, + ARRAY_SIZE(rt1011_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1011_dai_link_components; + link->num_codecs = ARRAY_SIZE(rt1011_dai_link_components); + link->init = rt1011_init; + link->ops = &rt1011_ops; +} + +void sof_rt1011_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1011_codec_confs; + card->num_configs = ARRAY_SIZE(rt1011_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h new file mode 100644 index 000000000000..87cb3812b926 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright(c) 2020 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with Realtek Codecs. + */ +#ifndef __SOF_REALTEK_COMMON_H +#define __SOF_REALTEK_COMMON_H + +#include <sound/soc.h> + +#define RT1011_CODEC_DAI "rt1011-aif" +#define RT1011_DEV0_NAME "i2c-10EC1011:00" +#define RT1011_DEV1_NAME "i2c-10EC1011:01" +#define RT1011_DEV2_NAME "i2c-10EC1011:02" +#define RT1011_DEV3_NAME "i2c-10EC1011:03" + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link); +void sof_rt1011_codec_conf(struct snd_soc_card *card); + +#endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 891f908659f5..8b1ca2da9bb9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -24,6 +24,7 @@ #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" #include "sof_maxim_common.h" +#include "sof_realtek_common.h" #define NAME_SIZE 32 @@ -41,10 +42,11 @@ #define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) -#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) -#define SOF_RT1015_SPEAKER_AMP_100FS BIT(14) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) +#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -741,6 +743,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].codecs = max98360a_component; links[id].num_codecs = ARRAY_SIZE(max98360a_component); links[id].init = speaker_codec_init; + } else if (sof_rt5682_quirk & + SOF_RT1011_SPEAKER_AMP_PRESENT) { + sof_rt1011_dai_link(&links[id]); } else { links[id].codecs = max98357a_component; links[id].num_codecs = ARRAY_SIZE(max98357a_component); @@ -851,6 +856,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) sof_max98373_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) + sof_rt1011_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -931,6 +938,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT1015_SPEAKER_AMP_100FS | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "tgl_rt1011_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1011_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; @@ -948,6 +964,7 @@ module_platform_driver(sof_audio) MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); @@ -955,3 +972,4 @@ MODULE_ALIAS("platform:jsl_rt5682_rt1015"); MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); +MODULE_ALIAS("platform:tgl_rt1011_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 9f243e60b95c..84cc6326979a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -310,6 +310,11 @@ static struct snd_soc_acpi_codecs tgl_max98373_amp = { .codecs = {"MX98373"} }; +static struct snd_soc_acpi_codecs tgl_rt1011_amp = { + .num_codecs = 1, + .codecs = {"10EC1011"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", @@ -335,6 +340,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "tgl_rt1011_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_rt1011_amp, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); -- 2.17.1

3 2

[PATCH] ASoC: amd: add missed checks for clk_get()
by Chuhong Yuan 03 Dec '20

03 Dec '20

cz_da7219_init() does not check the return value of clk_get(), while da7219_clk_enable() calls clk_set_rate() to dereference the pointers. Add checks to fix the problems. Fixes: bb24a31ed584 ("ASoC: AMD: Configure wclk and bclk of master codec") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a7702e64ec51..d5185cd3106b 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -74,7 +74,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) } da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + if (IS_ERR(da7219_dai_wclk)) + return PTR_ERR(da7219_dai_wclk); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); + if (IS_ERR(da7219_dai_bclk)) + return PTR_ERR(da7219_dai_bclk); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | -- 2.26.2

2 1

[PATCH 1/2] ASoC: fsl-asoc-card: Add support for si476x codec
by Shengjiu Wang 03 Dec '20

03 Dec '20

The si476x codec is used for FM radio function on i.MX6 auto board, it only supports recording function. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index a2dd3b6b7fec..f62f81ceab0d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -131,6 +131,13 @@ static const struct snd_soc_dapm_route audio_map_tx[] = { {"CPU-Playback", NULL, "ASRC-Playback"}, }; +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* 1st half -- Normal DAPM routes */ + {"CPU-Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"ASRC-Capture", NULL, "CPU-Capture"}, +}; + /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), @@ -653,6 +660,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); + } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { + codec_dai_name = "si476x-codec"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + priv->card.dapm_routes = audio_map_rx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -869,6 +881,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, { .compatible = "fsl,imx-audio-wm8524", }, + { .compatible = "fsl,imx-audio-si476x", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.27.0

3 3

[PATCH] ASoC: fsl_audmix: Remove unneeded data field
by Fabio Estevam 03 Dec '20

03 Dec '20

The .data field is only used to pass the string name to platform_device_register_data(). Pass the string name directly to make the code simpler. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/fsl_audmix.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 7ad5925772e8..8dc44dec7956 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -455,7 +455,6 @@ static const struct regmap_config fsl_audmix_regmap_config = { static const struct of_device_id fsl_audmix_ids[] = { { .compatible = "fsl,imx8qm-audmix", - .data = "imx-audmix", }, { /* sentinel */ } }; @@ -465,17 +464,9 @@ static int fsl_audmix_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct fsl_audmix *priv; - const char *mdrv; - const struct of_device_id *of_id; void __iomem *regs; int ret; - of_id = of_match_device(fsl_audmix_ids, dev); - if (!of_id || !of_id->data) - return -EINVAL; - - mdrv = of_id->data; - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; @@ -510,10 +501,10 @@ static int fsl_audmix_probe(struct platform_device *pdev) goto err_disable_pm; } - priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); + priv->pdev = platform_device_register_data(dev, "imx-audmix", 0, NULL, 0); if (IS_ERR(priv->pdev)) { ret = PTR_ERR(priv->pdev); - dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); + dev_err(dev, "failed to register platform: %d\n", ret); goto err_disable_pm; } -- 2.17.1

2 1

[PATCH 0/4] ASoC/SoundWire: add Realtek SoundWire SDCA codec
by Bard Liao 03 Dec '20

03 Dec '20

Add Realtek rt1316 and rt711-sdca vender-specific drivers plus some fixup for rt715-sdca driver. Jack Yu (2): ASoC/SoundWire: rt715-sdca: use rt715_sdca prefix for symbols ASoC/SoundWire: rt715-sdca: merge mute/volume operations Shuming Fan (2): ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver ASoC/SoundWire: rt711-sdca: Add RT711 SDCA vendor-specific driver sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt1316-sdw.c | 756 +++++++++++++++ sound/soc/codecs/rt1316-sdw.h | 115 +++ sound/soc/codecs/rt711-sdca-sdw.c | 424 +++++++++ sound/soc/codecs/rt711-sdca-sdw.h | 101 ++ sound/soc/codecs/rt711-sdca.c | 1482 +++++++++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 246 +++++ sound/soc/codecs/rt715-sdca-sdw.c | 12 +- sound/soc/codecs/rt715-sdca.c | 323 ++++--- sound/soc/codecs/rt715-sdca.h | 4 +- 11 files changed, 3316 insertions(+), 164 deletions(-) create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h -- 2.17.1

2 5

[PATCH v3 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

1 0

[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0

[PATCH v2 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

On 03 Dec 2020 07:33:03 +0900, Kuninori Morimoto wrote: >Hi Gyeongtaek > >Thank you for your patch > >> If stop by underrun and DPCM BE disconnection is run simultaneously, >> data abort can be occurred by the sequence below. >> >> /* In core X, running dpcm_be_dai_trigger() */ >> for_each_dpcm_be(fe, stream, dpcm) { >> /* In core Y, running dpcm_be_disconnect() */ >> spin_lock_irqsave(&fe->card->dpcm_lock, flags); >> list_del(&dpcm->list_be); >> list_del(&dpcm->list_fe); >> spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); >> kfree(dpcm); >> /* In core X, running dpcm_be_dai_trigger() */ >> struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory > >It is easy to read/understand if this code has alignment. > >> To prevent this situation, dpcm_lock should be acquired during >> iteration of dpcm list in dpcm_do_trigger(). >> >> Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> >> Cc: stable(a)vger.kernel.org >> --- > >Is this bugfix patch for some recent commit ? >If so, having Fixes: tag is helpful. > >Thank you for your help !! I tried to find out some change from the history. However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. So, I just fixed some miss from the commit comment and resend. If you have another suggestion or request, just let me know. And I have a question. This patch can’t be applied to stable branches directly by conflict. I found out following commits are needed to apply this patch into the 5.4 stable. ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code In this case, what should I do when stable maintainer requests resolving conflict to me? Should I send those patches to the maintainer? Or rewrite the patch on the latest of the each stable? Thank you! Gyeongtaek Lee > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

2 1

[PATCH v0] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by mdurnev＠gmail.com 03 Dec '20

03 Dec '20

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> --- sound/soc/sh/rcar/core.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6e670b3..8ca3fb7 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1428,8 +1428,73 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + + channel = io->converted_chan ? io->converted_chan : fe_channel; + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.7.4

4 4

[PATCH v2 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. /* In core X, running dpcm_be_dai_trigger() */ for_each_dpcm_be(fe, stream, dpcm) { /* In core Y, running dpcm_be_disconnect() */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); /* In core X, running dpcm_be_dai_trigger() */ struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

1 0

[PATCH 1/2] ASoC: dt-bindings: imx-hdmi: Add binding doc for hdmi machine driver
by Shengjiu Wang 03 Dec '20

03 Dec '20

Imx-hdmi is a new added machine driver for supporting hdmi devices on i.MX platforms. There is HDMI IP or external HDMI modules connect with SAI or AUD2HTX interface. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../bindings/sound/imx-audio-hdmi.yaml | 52 +++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml new file mode 100644 index 000000000000..d5474f83ac2c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-hdmi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio complex with HDMI + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-hdmi + - fsl,imx-audio-sii902x + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-cpu: + description: The phandle of an CPU DAI controller + + hdmi-out: + description: | + This is a boolean property. If present, the transmitting function + of HDMI will be enabled, indicating there's a physical HDMI out + connector or jack on the board or it's connecting to some other IP + block, such as an HDMI encoder or display-controller. + + hdmi-in: + description: | + This is a boolean property. If present, the receiving function of + HDMI will be enabled, indicating there is a physical HDMI in + connector/jack on the board. + +required: + - compatible + - model + - audio-cpu + +additionalProperties: false + +examples: + - | + sound-hdmi { + compatible = "fsl,imx-audio-hdmi"; + model = "audio-hdmi"; + audio-cpu = <&aud2htx>; + hdmi-out; + }; -- 2.27.0

3 3

[PATCH v2 0/5] regmap/SoundWire/ASoC: Add SoundWire SDCA support
by Bard Liao 03 Dec '20

03 Dec '20

The MIPI SoundWire Device Class standard will define audio functionality beyond the scope of the existing SoundWire 1.2 standard, which is limited to the bus and interface. The description is inspired by the USB Audio Class, with "functions", "entities", "control selectors", "audio clusters". The main difference with the USB Audio class is that the devices are typically on a motherboard and descriptors stored in platform firmware instead of being retrieved from the device. The current set of devices managed in this patchset are conformant with the SDCA 0.6 specification and require dedicated drivers since the descriptors and platform firmware specification is not complete at this time. They do however rely on the hierarchical addressing required by the SDCA standard. Future devices conformant with SDCA 1.0 should rely on a class driver. This series adds support for the hierarchical SDCA addressing and extends regmap. It then provides 3 codecs for RT711-sdca headset codec, RT1316 amplifier and RT715-scda microphone codec. Note that the release of this code before the formal adoption of the SDCA 1.0 specification was formally endorsed by the MIPI Board to make sure there is no delay for Linux-based support of this specification. v2: - rt715-sdca: Use rt715_sdca prefix to avoid compiling issue. - rt715-sdca: Merge multiple mute/volume operation into single mute/volume operation - rt711-sdca: Initial ret = 0 as it could be used uninitialized. Jack Yu (1): ASoC/SoundWire: rt715-sdca: First version of rt715 sdw sdca codec driver Pierre-Louis Bossart (2): soundwire: SDCA: add helper macro to access controls regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ Shuming Fan (2): ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver ASoC/SoundWire: rt711-sdca: Add RT711 SDCA vendor-specific driver drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 101 ++ include/linux/regmap.h | 35 + include/linux/soundwire/sdw_registers.h | 32 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/rt1316-sdw.c | 756 ++++++++++++ sound/soc/codecs/rt1316-sdw.h | 115 ++ sound/soc/codecs/rt711-sdca-sdw.c | 424 +++++++ sound/soc/codecs/rt711-sdca-sdw.h | 101 ++ sound/soc/codecs/rt711-sdca.c | 1482 +++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 246 ++++ sound/soc/codecs/rt715-sdca-sdw.c | 278 +++++ sound/soc/codecs/rt715-sdca-sdw.h | 170 +++ sound/soc/codecs/rt715-sdca.c | 947 +++++++++++++++ sound/soc/codecs/rt715-sdca.h | 124 ++ 17 files changed, 4843 insertions(+), 1 deletion(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h create mode 100644 sound/soc/codecs/rt715-sdca-sdw.c create mode 100644 sound/soc/codecs/rt715-sdca-sdw.h create mode 100644 sound/soc/codecs/rt715-sdca.c create mode 100644 sound/soc/codecs/rt715-sdca.h -- 2.17.1

5 15

[PATCH] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. /* In core X, running dpcm_be_dai_trigger() */ for_each_dpcm_be(fe, stream, dpcm) { /* In core Y, running dpcm_be_disconnect() */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); /* In core X, running dpcm_be_dai_trigger() */ struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_do_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

2 2

[PATCH] ASoC: codecs: lpass-va-macro: remove some dead code
by Dan Carpenter 02 Dec '20

02 Dec '20

The "decimator" variable is in the 0-7 range and it's unsigned so there is no need to check for negative values. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/lpass-va-macro.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 3e6bbef26dcb..91e6890d6efc 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -844,17 +844,10 @@ static int va_macro_hw_params(struct snd_pcm_substream *substream, for_each_set_bit(decimator, &va->active_ch_mask[dai->id], VA_MACRO_DEC_MAX) { - if (decimator >= 0) { - tx_fs_reg = CDC_VA_TX0_TX_PATH_CTL + - VA_MACRO_TX_PATH_OFFSET * decimator; - snd_soc_component_update_bits(component, tx_fs_reg, - 0x0F, tx_fs_rate); - } else { - dev_err(va_dev, - "%s: ERROR: Invalid decimator: %d\n", - __func__, decimator); - return -EINVAL; - } + tx_fs_reg = CDC_VA_TX0_TX_PATH_CTL + + VA_MACRO_TX_PATH_OFFSET * decimator; + snd_soc_component_update_bits(component, tx_fs_reg, 0x0F, + tx_fs_rate); } return 0; } -- 2.29.2

3 3

[RESEND PATCH 0/6] ASoC: mediatek: mt8192: support new machine rt1015p_rt5682
by Tzung-Bi Shih 02 Dec '20

02 Dec '20

The series supports new machines with rt1015p and rt5682. Reuses most of the code in mt8192-mt6359-rt1015-rt5682.c. The first 3 patches refactor the existing mt8192-mt6359-rt1015-rt5682.c. For easier to support newly added components later. The 4th patch fixes a typo in DT bindings document. The 5th patch proposes a new compatible string "mt8192_mt6359_rt1015p_rt5682" for machines with rt1015p and rt5682. The 6th patch reuses the existing machine driver to support machines with rt1015p and rt5682. Tzung-Bi Shih (6): ASoC: mediatek: mt8192: rename common symbols ASoC: mediatek: mt8192: extract rt1015_rt5682 specific DAI link ASoC: mediatek: mt8192: move rt1015_rt5682 specific data ASoC: dt-bindings: mt8192-mt6359: fix typo in the example ASoC: dt-bindings: mt8192-mt6359: add new compatible for using rt1015p ASoC: mediatek: mt8192: support rt1015p_rt5682 .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 6 +- sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 171 +++++++++++++----- 3 files changed, 129 insertions(+), 49 deletions(-) -- 2.29.2.454.gaff20da3a2-goog

2 7

[PATCH] ASoC: Intel: Boards: tgl_max98373: update TDM slot_width
by Ranjani Sridharan 02 Dec '20

02 Dec '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Speaker amp's SSP bclk configuration was changed in the topology file to be based on 12.288MHz and dai_ops->hw_params is based on s32le format. But, the TDM slot size remained set to 24 bits. This inconsistency created audible noises and needs to be corrected. This patch updates TDM slot width to 32. Fixes: bc7477fc2ab4 ("ASoC: Intel: Boards: tgl_max98373: Update TDM configuration in hw_params") Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index b6e63ea13d64..c2a9757181fe 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -49,11 +49,11 @@ static int max98373_hw_params(struct snd_pcm_substream *substream, for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) { /* DEV0 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 24); + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 32); } if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { /* DEV1 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 24); + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 32); } } return 0; -- 2.25.1

2 1

[PATCH] ASoC: rt715: remove unused parameter
by jack.yu＠realtek.com 02 Dec '20

02 Dec '20

From: Jack Yu <jack.yu(a)realtek.com> Remove unused parameter in rt715.h. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index d0d0fd2a6fdb..009a8266f606 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -207,7 +207,6 @@ struct sdw_stream_data { enum { RT715_AIF1, RT715_AIF2, - RT715_AIFS, }; #define RT715_POWER_UP_DELAY_MS 400 -- 2.29.0

2 1

How to interperet snd_pcm_htimestamp available frames?
by Eric Buddington 02 Dec '20

02 Dec '20

I am recording from multiple (cheap USB) mics, and trying to synchronize them. I am assuming that there will be a common minimum latency (reflecting an uncontested USB bus), with occasional spikes up - but not down. In my loop, I am calling snd_pcm_mmap_begin(), snd_pcm_htimestamp(), and snd__mmap_commit() in that order. My question is about the 'avail' value returned from htimestamp. It is documented as "Number of available frames when timestamp was grabbed". Does this mean frames since the last frame consumed with _mmap_commit()? Is there a possibility of there being no timestamp, or one where 'avail' refers to some older state? Thanks for any insight. -Eric

2 1

[PATCH] pcm: set the snd_pcm_ioplug_status() tstamp field
by Jonas Holmberg 02 Dec '20

02 Dec '20

Set the status tstamp field so that it can be accessed with snd_pcm_status_get_htstamp(). Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_ioplug.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a437ca32..9b1b8ac3 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -115,6 +115,7 @@ static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) snd_pcm_ioplug_hw_ptr_update(pcm); status->state = io->data->state; status->trigger_tstamp = io->trigger_tstamp; + gettimestamp(&status->tstamp, pcm->tstamp_type); status->avail = snd_pcm_mmap_avail(pcm); status->avail_max = io->avail_max; return 0; -- 2.26.2

2 2

[PATCH v1] ASoC: Intel: catpt: Replace open coded variant of resource_intersection()
by Andy Shevchenko 02 Dec '20

02 Dec '20

Since we have resource_intersection() helper, let's utilize it here. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- Promised clean up for recently introduced helper. This has dependency to the patches currently in linux-pm tree. Other than that everything else is already in upstream. Hence, logically it's better to push thru Rafael's tree than wait one more cycle. sound/soc/intel/catpt/core.h | 11 ----------- sound/soc/intel/catpt/loader.c | 2 +- 2 files changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/intel/catpt/core.h b/sound/soc/intel/catpt/core.h index 0f53a0d43254..a64a0a77dcb7 100644 --- a/sound/soc/intel/catpt/core.h +++ b/sound/soc/intel/catpt/core.h @@ -22,17 +22,6 @@ void catpt_sram_free(struct resource *sram); struct resource * catpt_request_region(struct resource *root, resource_size_t size); -static inline bool catpt_resource_overlapping(struct resource *r1, - struct resource *r2, - struct resource *ret) -{ - if (!resource_overlaps(r1, r2)) - return false; - ret->start = max(r1->start, r2->start); - ret->end = min(r1->end, r2->end); - return true; -} - struct catpt_ipc_msg { union { u32 header; diff --git a/sound/soc/intel/catpt/loader.c b/sound/soc/intel/catpt/loader.c index 40c22e4bb263..ff7b8f0d34ac 100644 --- a/sound/soc/intel/catpt/loader.c +++ b/sound/soc/intel/catpt/loader.c @@ -267,7 +267,7 @@ static int catpt_restore_fwimage(struct catpt_dev *cdev, r2.start = off; r2.end = r2.start + info->size - 1; - if (!catpt_resource_overlapping(&r2, &r1, &common)) + if (!resource_intersection(&r2, &r1, &common)) continue; /* calculate start offset of common data area */ off = common.start - r1.start; -- 2.29.2

4 8

[PATCH 1/2] ALSA: ppc: drop if block with always false condition
by Uwe Kleine-König 02 Dec '20

02 Dec '20

The remove callback is only called for devices that were probed successfully before. As the matching probe function cannot complete without error if dev->match_id != PS3_MATCH_ID_SOUND, we don't have to check this here. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/ppc/snd_ps3.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 58bb49fff184..6ab796a5d936 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -1053,8 +1053,6 @@ static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev) { int ret; pr_info("%s:start id=%d\n", __func__, dev->match_id); - if (dev->match_id != PS3_MATCH_ID_SOUND) - return -ENXIO; /* * ctl and preallocate buffer will be freed in -- 2.29.2

6 15

[PATCH] soundwire: master: use pm_runtime_set_active() on add
by Bard Liao 02 Dec '20

02 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The 'master' device acts as a glue layer used during bus initialization only, and it needs to be 'transparent' for pm_runtime management. Its behavior should be that it becomes active when one of its children becomes active, and suspends when all of its children are suspended. In our tests on Intel platforms, we routinely see these sort of warnings on the initial boot: [ 21.447345] rt715 sdw:3:25d:715:0: runtime PM trying to activate child device sdw:3:25d:715:0 but parent (sdw-master-3) is not active This is root-caused to a missing setup to make the device 'active' on probe. Since we don't want the device to remain active forever after the probe, the autosuspend configuration is also enabled at the end of the probe - the device will actually autosuspend only in the case where there are no devices physically attached. In practice, the master device will suspend when all its children are no longer active. Fixes: bd84256e86ecf ('soundwire: master: enable pm runtime') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/master.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 3488bb824e84..9b05c9e25ebe 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -8,6 +8,15 @@ #include <linux/soundwire/sdw_type.h> #include "bus.h" +/* + * The 3s value for autosuspend will only be used if there are no + * devices physically attached on a bus segment. In practice enabling + * the bus operation will result in children devices become active and + * the master device will only suspend when all its children are no + * longer active. + */ +#define SDW_MASTER_SUSPEND_DELAY_MS 3000 + /* * The sysfs for properties reflects the MIPI description as given * in the MIPI DisCo spec @@ -154,7 +163,12 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, bus->dev = &md->dev; bus->md = md; + pm_runtime_set_autosuspend_delay(&bus->md->dev, SDW_MASTER_SUSPEND_DELAY_MS); + pm_runtime_use_autosuspend(&bus->md->dev); + pm_runtime_mark_last_busy(&bus->md->dev); + pm_runtime_set_active(&bus->md->dev); pm_runtime_enable(&bus->md->dev); + pm_runtime_idle(&bus->md->dev); device_register_err: return ret; } -- 2.17.1

3 5

Sound BlasterX AE-5 not working
by Oliver Z. 02 Dec '20

02 Dec '20

Since a couple of month i am trying to get that sound card working. First it seemed to be a pulseaudio problem but now it looks like its an alsa/kernel issue. with some help from #alsa we figured out its the same problem as mentioned here: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/935 with everything connected: > jacks_do card=0 card=1 numid=51,iface=CARD,name='Line Out CLFE Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=49,iface=CARD,name='Line Out Front Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=50,iface=CARD,name='Line Out Surround Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=48,iface=CARD,name='Line Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=46,iface=CARD,name='Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=47,iface=CARD,name='Mic Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=54,iface=CARD,name='SPDIF In Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=52,iface=CARD,name='SPDIF Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=53,iface=CARD,name='SPDIF Phantom Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on card=2 numid=25,iface=CARD,name='HDMI/DP,pcm=10 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=31,iface=CARD,name='HDMI/DP,pcm=11 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=37,iface=CARD,name='HDMI/DP,pcm=12 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=1,iface=CARD,name='HDMI/DP,pcm=3 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off line out jack removed > jacks_do card=0 card=1 numid=51,iface=CARD,name='Line Out CLFE Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=49,iface=CARD,name='Line Out Front Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=50,iface=CARD,name='Line Out Surround Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=48,iface=CARD,name='Line Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=46,iface=CARD,name='Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=47,iface=CARD,name='Mic Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=off numid=54,iface=CARD,name='SPDIF In Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=52,iface=CARD,name='SPDIF Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=53,iface=CARD,name='SPDIF Phantom Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on card=2 numid=25,iface=CARD,name='HDMI/DP,pcm=10 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=31,iface=CARD,name='HDMI/DP,pcm=11 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=37,iface=CARD,name='HDMI/DP,pcm=12 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=1,iface=CARD,name='HDMI/DP,pcm=3 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off and line out plugged in again > jacks_do card=0 card=1 numid=51,iface=CARD,name='Line Out CLFE Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=49,iface=CARD,name='Line Out Front Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=50,iface=CARD,name='Line Out Surround Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=48,iface=CARD,name='Line Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=46,iface=CARD,name='Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=47,iface=CARD,name='Mic Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=54,iface=CARD,name='SPDIF In Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=52,iface=CARD,name='SPDIF Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=53,iface=CARD,name='SPDIF Phantom Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on card=2 numid=25,iface=CARD,name='HDMI/DP,pcm=10 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=31,iface=CARD,name='HDMI/DP,pcm=11 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=37,iface=CARD,name='HDMI/DP,pcm=12 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=1,iface=CARD,name='HDMI/DP,pcm=3 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off i would love to have this soundcard working. i had to buy a cheap usb soundcard now to get my headset working on linux (fedora 33). on windows it works as expected

1 0

Asoc: Intel: SST (CHT) regression in asoc/for-5.11
by Hans de Goede 02 Dec '20

02 Dec '20

Hi All, To test the code to dynamically switch between SST/SOF support on BYT/CHT from the kernel commandline I merged: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/log/?h=fo… Into my personal tree (mostly Linus' master + some pending patches from myself). After this I was getting the following errors in dmesg when using sound on a Medion E2228T laptop with a CHT SoC + NAU8824 codec: [ 53.805205] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0 [ 53.805479] intel_sst_acpi 808622A8:00: fw returned err -16 [ 53.806281] sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16 [ 54.829548] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0 [ 54.829596] intel_sst_acpi 808622A8:00: fw returned err -16 [ 54.829668] sst-mfld-platform sst-mfld-platform: ASoC: POST_PMD: media0_out event failed: - [ 55.853230] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0 [ 55.853244] intel_sst_acpi 808622A8:00: fw returned err -16 [ 55.853269] sst-mfld-platform sst-mfld-platform: ASoC: POST_PMD: codec_out0 mix 0 event fai [ 56.876435] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0 [ 56.876481] intel_sst_acpi 808622A8:00: fw returned err -16 [ 56.876563] sst-mfld-platform sst-mfld-platform: ASoC: POST_PMD: media0_out mix 0 event fai [ 61.847455] intel_sst_acpi 808622A8:00: FW sent error response 0x40015 [ 61.847564] intel_sst_acpi 808622A8:00: fw returned err -1 [ 61.847659] sst-mfld-platform sst-mfld-platform: ASoC: error at snd_soc_dai_startup on ssp2 [ 61.847722] SSP2-Codec: ASoC: BE open failed -1 [ 61.847754] Audio Port: ASoC: failed to start some BEs -1 [ 61.847786] intel_sst_acpi 808622A8:00: FW sent error response 0x40006 [ 64.301284] intel_sst_acpi 808622A8:00: FW sent error response 0x90001 [ 64.301545] intel_sst_acpi 808622A8:00: not suspending FW!!, Err: -2 Dropping the asoc/for-5.11 merge and just cherry-picking Pierre-Louis' changes for the dynamic switching makes these go away. So this seems to be caused by other changes in asoc/for-5.11. So any clues where to start looking for this, or should I just bisect this? Regards, Hans

4 7

Re: linux-next: Tree for Dec 1 (sound: SND_SOC_ADAU1372_SPI)
by Randy Dunlap 01 Dec '20

01 Dec '20

On 12/1/20 2:03 AM, Stephen Rothwell wrote: > Hi all, > > Changes since 20201130: > on i386 or x86_64: WARNING: unmet direct dependencies detected for REGMAP_SPI Depends on [n]: SPI [=n] Selected by [m]: - SND_SOC_ADAU1372_SPI [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] Full x86_64 randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

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module-alsa-card.c: Failed to find a working profile.
by GitHub issues - opened 01 Dec '20

01 Dec '20

alsa-project/alsa-lib issue #104 was opened from mhwombat: I can get audio through my headphones, but not through my speakers. Basic checks: - Speakers work on another machine - Speakers are plugged in and turned on - Headphones are unplugged when using speakers - My user account is in the audio group. - Nothing is muted in alsamixer I think the cause might be this `module-alsa-card.c: Failed to find a working profile.` ``` $ journalctl --user -u pulseaudio | tail -n 20 -- Reboot -- Nov 28 22:35:15 nixos systemd[2142]: Starting Sound Service... Nov 28 22:35:15 nixos pulseaudio[2160]: E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. Nov 28 22:35:15 nixos pulseaudio[2160]: E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="pci-0000_22_00.4" card_name="alsa_card.pci-0000_22_00.4" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. Nov 28 22:35:17 nixos pulseaudio[2160]: E: [pulseaudio] module-jackdbus-detect.c: Unable to contact D-Bus session bus: org.freedesktop.DBus.Error.NotSupported: Unable to autolaunch a dbus-daemon without a $DISPLAY for X11 Nov 28 22:35:17 nixos pulseaudio[2160]: E: [pulseaudio] module.c: Failed to load module "module-jackdbus-detect" (argument: "channels=2"): initialization failed. Nov 28 22:35:17 nixos pulseaudio[2160]: W: [pulseaudio] server-lookup.c: Unable to contact D-Bus: org.freedesktop.DBus.Error.NotSupported: Unable to autolaunch a dbus-daemon without a $DISPLAY for X11 Nov 28 22:35:17 nixos pulseaudio[2160]: W: [pulseaudio] main.c: Unable to contact D-Bus: org.freedesktop.DBus.Error.NotSupported: Unable to autolaunch a dbus-daemon without a $DISPLAY for X11 Nov 28 22:35:17 nixos systemd[2142]: Started Sound Service. Nov 28 22:35:17 nixos pulseaudio[2160]: E: [pulseaudio] bluez5-util.c: GetManagedObjects() failed: org.freedesktop.DBus.Error.ServiceUnknown: The name org.bluez was not provided by any .service files Nov 29 17:24:50 nixos systemd[2142]: pulseaudio.service: Current command vanished from the unit file, execution of the command list won't be resumed. Nov 29 17:25:00 nixos pulseaudio[2160]: E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. Nov 29 17:25:00 nixos pulseaudio[2160]: E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="pci-0000_22_00.4" card_name="alsa_card.pci-0000_22_00.4" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. Nov 29 17:25:00 nixos pulseaudio[2160]: E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. Nov 29 17:25:00 nixos pulseaudio[2160]: E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="pci-0000_22_00.4" card_name="alsa_card.pci-0000_22_00.4" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. Nov 29 17:25:01 nixos pulseaudio[2160]: E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. Nov 29 17:25:01 nixos pulseaudio[2160]: E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="pci-0000_22_00.4" card_name="alsa_card.pci-0000_22_00.4" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. Dec 01 10:53:45 nixos pulseaudio[2160]: E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Dec 01 10:53:45 nixos pulseaudio[2160]: E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Dec 01 10:53:45 nixos pulseaudio[2160]: E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` Here is the [output of alsa-info_alsa-info.sh](https://alsa-project.org/db/?f=c5a24a41c9ab6f2b3203…. Issue URL : https://github.com/alsa-project/alsa-lib/issues/104 Repository URL: https://github.com/alsa-project/alsa-lib

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Re: [RFC] ASoC: Add compatible for mt6359-sound device
by Mark Brown 01 Dec '20

01 Dec '20

On Tue, Dec 01, 2020 at 01:41:33PM +0800, Shane Chien wrote: > However, if the device is not given a comaptible, > of_node of struct device is null. I cannot use > devm_iio_channel_get such iio interface to get > auxadc value from iio channel. Because during > using devm_iio_channel_get, of_node of mt6359-sound is a > input parameter of of_iio_channel_get_by_name. > If the of_node is null, devm_iio_channel_get will > eventually return ENODEV error. I would expect the IIO channel to be requestable using the top level device for the MFD - part of the deal here is that the function drivers for the MFD know they're part of the MFD so can look at their parent device for some things (eg, this is how regmaps are normally obtained).

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