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[PATCH AUTOSEL 5.15 04/26] ASoC: SOF: Intel: pci-tgl: add ADL-N support
by Sasha Levin 27 Dec '21

27 Dec '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit cd57eb3c403cb864e5558874ecd57dd954a5a7f7 ] Add PCI DID for Intel AlderLake-N. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20211203171542.1021399-1-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/pci-tgl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index d04ce84fe7cc2..665add8a3029b 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -119,6 +119,8 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&adl_desc}, { PCI_DEVICE(0x8086, 0x51cc), /* ADL-M */ .driver_data = (unsigned long)&adl_desc}, + { PCI_DEVICE(0x8086, 0x54c8), /* ADL-N */ + .driver_data = (unsigned long)&adl_desc}, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); -- 2.34.1

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[RFC 09/32] sound: Kconfig: add HAS_IOPORT dependencies
by Niklas Schnelle 27 Dec '21

27 Dec '21

In a future patch HAS_IOPORT=n will result in inb()/outb() and friends not being declared. We thus need to add HAS_IOPORT as dependency for those drivers using them. Co-developed-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Niklas Schnelle <schnelle(a)linux.ibm.com> --- sound/drivers/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index ca4cdf666f82..4b5ba916d93e 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -128,6 +128,7 @@ config SND_VIRMIDI config SND_MTPAV tristate "MOTU MidiTimePiece AV multiport MIDI" + depends on HAS_IOPORT select SND_RAWMIDI help To use a MOTU MidiTimePiece AV multiport MIDI adapter @@ -152,6 +153,7 @@ config SND_MTS64 config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" + depends on HAS_IOPORT select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here @@ -167,6 +169,7 @@ config SND_SERIAL_U16550 config SND_MPU401 tristate "Generic MPU-401 UART driver" + depends on HAS_IOPORT select SND_MPU401_UART help Say Y here to include support for MIDI ports compatible with -- 2.32.0

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a52: frozen pavumeter
by GitHub issues - edited 27 Dec '21

27 Dec '21

alsa-project/alsa-plugins issue #42 was edited from quequotion: I had been [thinking that pavumeter was just broken](https://github.com/nekohayo/pavumeter/issues/1), because this has been happening to me for a long time (well before the changes between 1.2.5 and 1.2.6), but now I realize that I hadn't tested against stereo mode in just as long. When using the a52 plugin, pavumeter is frozen as soon as it opens: if audio was playing, levels will show at exactly where they were at that moment and never change; if audio was not playing, levels will never appear because the interface is frozen before there were any. I am curious if this, #37 and #32 are not related. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/42 Repository URL: https://github.com/alsa-project/alsa-plugins

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a52: frozen pavumeter
by GitHub issues - opened 27 Dec '21

27 Dec '21

alsa-project/alsa-plugins issue #42 was opened from quequotion: I had been thinking that pavumeter was just broken, because this has been happening to me for a long time (well before the changes between 1.2.5 and 1.2.6), but now I realize that I hadn't tested against stereo mode in just as long. When using the a52 plugin, pavumeter is frozen as soon as it opens: if audio was playing, levels will show at exactly where they were at that moment and never change; if audio was not playing, levels will never appear because the interface is frozen before there were any. I am curious if this, #37 and #32 are not related. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/42 Repository URL: https://github.com/alsa-project/alsa-plugins

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After Update: No more sound. Downgrading solved it.
by GitHub issues - edited 27 Dec '21

27 Dec '21

alsa-project/alsa-ucm-conf issue #131 was edited from aljazzair: Hello. I am on Manjaro Linux, an Arch derivative. Recently, after a system update, sound was gone. After a couple of hours investigation, I narrowed it down to an update of alsa-ucm-conf. Basically after downgrading 1.2.6.3-0 => 1.2.5.1-1, my problem is gone. Symptom: With 1.2.6.3-0, there was no sound. Running aplay -l listed my hardware. Also I was able to hear sound from my monitor using the following command: > aplay -D plughw:0,7 /usr/share/sounds/alsa/Front_Right.wav But pactl list sinks came back empty. Pavucontrol showed no output devices either. Rebooting several times, editing various config files did not help. I looked at the list of recent upgrades and saw alsa-ucm-conf in the logs. Downgrading solved my issue. Distribution: > Linux homelab 5.13.19-2-MANJARO #1 SMP PREEMPT Sun Sep 19 21:31:53 UTC 2021 x86_64 GNU/Linux List of PLAYBACK Hardware Devices from aplay -l: > card 0: PCH [HDA Intel PCH], device 0: ALC1220 Analog [ALC1220 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 1: ALC1220 Digital [ALC1220 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > Subdevices: 1/1 > Subdevice #0: subdevice #0 Sound managed with: > pipewire --version > pipewire > Compiled with libpipewire 0.3.40 > Linked with libpipewire 0.3.40 > I am posting this in order to document what happened in case someone has a similar issue, and searches google for hints. Also to inform the developer, maybe something can be done to avoid this. Thank you. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/131 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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After Update: No more sound. Downgrading solved it.
by GitHub issues - edited 27 Dec '21

27 Dec '21

alsa-project/alsa-ucm-conf issue #131 was edited from aljazzair: Hello. I am on Manjaro Linux, an Arch derivative. Recently, after a system update, sound was gone. After a couple of hours investigation, I narrowed it down to an update of alsa-ucm-conf. Basically after downgrading 1.2.6.3-0 => 1.2.5.1-1, my problem is gone. Symptom: With 1.2.6.3-0, there was no sound. Running aplay -l listed my hardware. Also I was able to hear sound from my monitor using the following command: > aplay -D plughw:0,7 /usr/share/sounds/alsa/Front_Right.wav But pactl list sinks came back empty. Pavucontrol showed no output devices either. Rebooting several times, editing various config files did not help. I looked at the list of recent upgrades and saw alsa-ucm-conf in the logs. Downgrading solved my issue. Distribution: > Linux homelab 5.13.19-2-MANJARO #1 SMP PREEMPT Sun Sep 19 21:31:53 UTC 2021 x86_64 GNU/Linux List of PLAYBACK Hardware Devices from aplay -l: > card 0: PCH [HDA Intel PCH], device 0: ALC1220 Analog [ALC1220 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 1: ALC1220 Digital [ALC1220 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > Subdevices: 1/1 > Subdevice #0: subdevice #0 Sound managed with: > pipewire --version > pipewire > Compiled with libpipewire 0.3.40 > Linked with libpipewire 0.3.40 > I am posting this in order to document what happened in case someone has the same issue, and searches google for hints. Also to inform the developper, maybe something can be done to avoid this. Thank you. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/131 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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After Update: No more sound. Downgrading solved it.
by GitHub issues - edited 27 Dec '21

27 Dec '21

alsa-project/alsa-ucm-conf issue #131 was edited from aljazzair: Hello. I am on Manjaro Linux, an Arch derivative. Recently, after a system update, sound was gone. After a couple of hours investigation, I narrowed it down to an update of alsa-ucm-conf. Basically after downgrading 1.2.6.3-0 => 1.2.5.1-1, my problem is gone. Symptom: With 1.2.6.3-0, there was no sound. Running aplay -l listed my hardware. Also I was able to hear sound from my monitor using the following command: > aplay -D plughw:0,7 /usr/share/sounds/alsa/Front_Right.wav But pactl list sinks came back empty. Pavucontrol showed no output devices either. Rebooting several times, editing various config files did not help. I looked at the list of recent upgrades and saw alsa-ucm-conf in the logs. Downgrading solved my issue. Distribution: > Linux homelab 5.13.19-2-MANJARO #1 SMP PREEMPT Sun Sep 19 21:31:53 UTC 2021 x86_64 GNU/Linux List of PLAYBACK Hardware Devices from aplay -l: > card 0: PCH [HDA Intel PCH], device 0: ALC1220 Analog [ALC1220 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 1: ALC1220 Digital [ALC1220 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > Subdevices: 1/1 > Subdevice #0: subdevice #0 Sound managed with: > pipewire --version > pipewire > Compiled with libpipewire 0.3.40 > Linked with libpipewire 0.3.40 > I am posting this in order to document what happened in case someone has the same issue, and searches google. Also to inform the developper, maybe something can be done to avoid this. Thank you. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/131 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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After Update: No more sound. Downgrading solved it.
by GitHub issues - edited 27 Dec '21

27 Dec '21

alsa-project/alsa-ucm-conf issue #131 was edited from aljazzair: Hello. I am on Manjaro Linux, an Arch derivative. Recently, after a system update, sound was gone. After a couple of hours investigation, I narrowed it down to an update of alsa-ucm-conf. Basically after downgrading 1.2.6.3-0 => 1.2.5.1-1, my problem is gone. I Symptom: With 1.2.6.3-0, there was no sound. Running aplay -l listed my hardware. Also I was able to hear sound from my monitor using the following command: > aplay -D plughw:0,7 /usr/share/sounds/alsa/Front_Right.wav But pactl list sinks came back empty. Pavucontrol showed no output devices either. Rebooting several times, editing various config files did not help. I looked at the list of recent upgrades and saw alsa-ucm-conf in the logs. Downgrading solved my issue. Distribution: > Linux homelab 5.13.19-2-MANJARO #1 SMP PREEMPT Sun Sep 19 21:31:53 UTC 2021 x86_64 GNU/Linux List of PLAYBACK Hardware Devices from aplay -l: > card 0: PCH [HDA Intel PCH], device 0: ALC1220 Analog [ALC1220 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 1: ALC1220 Digital [ALC1220 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > Subdevices: 1/1 > Subdevice #0: subdevice #0 Sound managed with: > pipewire --version > pipewire > Compiled with libpipewire 0.3.40 > Linked with libpipewire 0.3.40 > I am posting this in order to document what happened in case someone has the same issue, and searches google. Also to inform the developper, maybe something can be done to avoid this. Thank you. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/131 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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After Update: No more sound. Downgrading solved it.
by GitHub issues - opened 27 Dec '21

27 Dec '21

alsa-project/alsa-ucm-conf issue #131 was opened from aljazzair: Hello. I am on Manjaro Linux, an Arch derivative. Recently, after a system update, sound was gone. After a couple of hours investigation, I narrowed it down to an update of alsa-ucm-conf. Basically after downgrading 1.2.6.3-0 => 1.2.5.1-1, my problem is gone. Symptom: With 1.2.6.3-0, there was no sound. Running aplay -l listed my hardware. Also I was able to hear sound from my monitor using the following command: > aplay -D plughw:0,7 /usr/share/sounds/alsa/Front_Right.wav But pactl list sinks came back empty. Pavucontrol showed no output devices either. Rebooting several times, editing various config files did not help. I looked at the list of recent upgrades and saw alsa-ucm-conf in the logs. Downgrading solved my issue. Distribution: > Linux netsrvr 5.13.19-2-MANJARO #1 SMP PREEMPT Sun Sep 19 21:31:53 UTC 2021 x86_64 GNU/Linux List of PLAYBACK Hardware Devices from aplay -l: > card 0: PCH [HDA Intel PCH], device 0: ALC1220 Analog [ALC1220 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 1: ALC1220 Digital [ALC1220 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > Subdevices: 1/1 > Subdevice #0: subdevice #0 Sound managed with: > pipewire --version > pipewire > Compiled with libpipewire 0.3.40 > Linked with libpipewire 0.3.40 > I am posting this in order to document what happened in case someone has the same issue, and searches google. Also to inform the developper, maybe something can be done to avoid this. Thank you. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/131 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: Use ALC285_FIXUP_HP_GPIO_LED on another HP laptop
by Kai-Heng Feng 25 Dec '21

25 Dec '21

The audio mute and mic mute LEDs don't work, so use the quirk to make them work. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- 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 28255e752c4a1..344741869281f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8730,6 +8730,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89c3, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.33.1

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[PATCH] ALSA: hda/realtek: Add speaker fixup for some Yoga 15ITL5 devices
by arsgeiger＠gmail.com 25 Dec '21

25 Dec '21

From: Arie Geiger <arsgeiger(a)gmail.com> This patch adds another possible subsystem ID for the ALC287 used by the Lenovo Yoga 15ITL5. It uses the same initalization as the others. This patch has been tested and works for my device. Signed-off-by: Arie Geiger <arsgeiger(a)gmail.com> --- 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 3599f4c85ebf..188cbcde6e49 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8901,6 +8901,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), -- 2.34.1

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[PATCH 1/2] ALSA: hda: Add AlderLake-N PCI ID
by Kai Vehmanen 25 Dec '21

25 Dec '21

Add HD Audio PCI ID for Intel AlderLake-N. Add rules to snd_intel_dsp_find_config() to choose DSP-based SOF driver for ADL-N systems with PCH-DMIC or Soundwire codecs, and plain HDA driver for the rest (DSP not used). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b5bfa94bcb89..e035cecb37f7 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -374,6 +374,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51cc, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x54c8, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b98d7975adea..cdd4a58244d7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2496,6 +2496,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-M */ { PCI_DEVICE(0x8086, 0x51cc), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-N */ + { PCI_DEVICE(0x8086, 0x54c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, base-commit: 74a50bd2899fdc9b6b2f2eb9624551e4aa795ad8 -- 2.34.1

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[PATCH] ALSA: hda/hdmi: Disable silent stream on GLK
by Ville Syrjala 25 Dec '21

25 Dec '21

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> The silent stream stuff recurses back into i915 audio component .get_power() from the .pin_eld_notify() hook. On GLK this will deadlock as i915 may already be holding the relevant modeset locks during .pin_eld_notify() and the GLK audio vs. CDCLK workaround will try to grab the same locks from .get_power(). Until someone comes up with a better fix just disable the silent stream support on GLK. Cc: stable(a)vger.kernel.org Cc: Harsha Priya <harshapriya.n(a)intel.com> Cc: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Closes: https://gitlab.freedesktop.org/drm/intel/-/issues/2623 Fixes: 951894cf30f4 ("ALSA: hda/hdmi: Add Intel silent stream support") Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 415701bd10ac..ffcde7409d2a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2947,7 +2947,8 @@ static int parse_intel_hdmi(struct hda_codec *codec) /* Intel Haswell and onwards; audio component with eld notifier */ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, - const int *port_map, int port_num, int dev_num) + const int *port_map, int port_num, int dev_num, + bool send_silent_stream) { struct hdmi_spec *spec; int err; @@ -2980,7 +2981,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, * Enable silent stream feature, if it is enabled via * module param or Kconfig option */ - if (enable_silent_stream) + if (send_silent_stream) spec->send_silent_stream = true; return parse_intel_hdmi(codec); @@ -2988,12 +2989,18 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, static int patch_i915_hsw_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x08, NULL, 0, 3); + return intel_hsw_common_init(codec, 0x08, NULL, 0, 3, + enable_silent_stream); } static int patch_i915_glk_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3); + /* + * Silent stream calls audio component .get_power() from + * .pin_eld_notify(). On GLK this will deadlock in i915 due + * to the audio vs. CDCLK workaround. + */ + return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3, false); } static int patch_i915_icl_hdmi(struct hda_codec *codec) @@ -3004,7 +3011,8 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) */ static const int map[] = {0x0, 0x4, 0x6, 0x8, 0xa, 0xb}; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3); + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3, + enable_silent_stream); } static int patch_i915_tgl_hdmi(struct hda_codec *codec) @@ -3016,7 +3024,8 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; int ret; - ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4, + enable_silent_stream); if (!ret) { struct hdmi_spec *spec = codec->spec; -- 2.32.0

3 2

Re: [PATCH] ALSA: hda: use swap() to make code cleaner
by Takashi Iwai 25 Dec '21

25 Dec '21

On Sat, 18 Dec 2021 02:54:16 +0100, davidcomponentone(a)gmail.com wrote: > > From: Yang Guang <yang.guang5(a)zte.com.cn> > > Use the macro 'swap()' defined in 'include/linux/minmax.h' to avoid > opencoding it. > > Reported-by: Zeal Robot <zealci(a)zte.com.cn> > Signed-off-by: David Yang <davidcomponentone(a)gmail.com> > Signed-off-by: Yang Guang <yang.guang5(a)zte.com.cn> Applied now. Thanks. Takashi

1 0

[PATCH v1 0/2] kselftest: alsa: Small enhancements to mixer-test
by Mark Brown 25 Dec '21

25 Dec '21

These two patches improve the mixer test, checking that the default values for enums are valid and extending them to cover all the values for multi-value controls, not just the first one. It also factors out the validation that values are OK for controls for future reuse. Mark Brown (2): kselftest: alsa: Factor out check that values meet constraints kselftest: alsa: Validate values read from enumerations tools/testing/selftests/alsa/mixer-test.c | 158 ++++++++++++++-------- 1 file changed, 99 insertions(+), 59 deletions(-) base-commit: b73dad806533cad55df41a9c0349969b56d4ff7f -- 2.30.2

3 7

[PATCH 0/2] ALSA/ASoC: hdac_ext cleanups
by Pierre-Louis Bossart 25 Dec '21

25 Dec '21

One function moved since it doesn't use the hdac_ext parts, and renaming of variables to identify when hdac_stream and hdac_ext_streams are used. No functionality change. Pierre-Louis Bossart (2): ALSA/ASoC: hda: move/rename snd_hdac_ext_stop_streams to hdac_stream.c ALSA: HDA: hdac_ext_stream: use consistent prefixes for variables include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 27 ++-- sound/hda/ext/hdac_ext_stream.c | 216 +++++++++++++++----------------- sound/hda/hdac_stream.c | 16 +++ sound/soc/intel/skylake/skl.c | 4 +- 5 files changed, 132 insertions(+), 132 deletions(-) -- 2.25.1

3 4

[PATCH v2 0/7] ASoC/SoundWire: improve suspend flows and use set_stream() instead of set_tdm_slots() for HDAudio
by Bard Liao 24 Dec '21

24 Dec '21

This series contains three topics. 1. SoundWire: Intel: remove pdm support 2. ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire 3. ASoC/SOF/SoundWire: fix suspend-resume on pause with dynamic pipelines The topics are independent but the changes are dependent. So please allow me to send them in one series. The code is basically not changed, so I still take Vinod Koul's Ack. v2: - ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire: rename .set_sdw_stream to .set_stream and .get_sdw_stream to .get_stream in intel_pdm_dai_ops. Pierre-Louis Bossart (6): ASOC: SOF: Intel: use snd_soc_dai_get_widget() ASoC/soundwire: intel: simplify callbacks for params/hw_free ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire ASoC: Intel/SOF: use set_stream() instead of set_tdm_slots() for HDAudio soundwire: intel: remove unnecessary init soundwire: intel: remove PDM support Ranjani Sridharan (1): soundwire: intel: improve suspend flows drivers/soundwire/cadence_master.c | 36 +--- drivers/soundwire/cadence_master.h | 14 +- drivers/soundwire/intel.c | 253 ++++++++++++++-------------- drivers/soundwire/qcom.c | 8 +- drivers/soundwire/stream.c | 4 +- include/linux/soundwire/sdw_intel.h | 4 +- include/sound/soc-dai.h | 32 ++-- sound/soc/codecs/hdac_hda.c | 22 +-- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdca.c | 2 +- sound/soc/codecs/rt715.c | 2 +- sound/soc/codecs/sdw-mockup.c | 2 +- sound/soc/codecs/wcd938x.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 6 +- sound/soc/intel/skylake/skl-pcm.c | 7 +- sound/soc/qcom/sdm845.c | 4 +- sound/soc/qcom/sm8250.c | 4 +- sound/soc/sof/intel/hda-dai.c | 7 +- sound/soc/sof/intel/hda.c | 12 +- 26 files changed, 190 insertions(+), 247 deletions(-) -- 2.17.1

2 8

[PATCH 0/4] ASoC: mediatek: fix device_node leak
by Tzung-Bi Shih 24 Dec '21

24 Dec '21

The series is a follow up series. The 1st and 2nd patch follow [1] to fix the device_node leak. The 3rd patch follows [2] to reduce the log verbosity. The 4th patch cleans the device_node related code on MT8195. In order to align to previous platforms. [1]: https://patchwork.kernel.org/project/alsa-devel/patch/20211214040028.299262… [2]: https://patchwork.kernel.org/project/alsa-devel/patch/20211220093408.207206… Tzung-Bi Shih (4): ASoC: mediatek: mt8173: fix device_node leak ASoC: mediatek: mt8183: fix device_node leak ASoC: mediatek: mt8173: reduce log verbosity in probe() ASoC: mediatek: mt8195: release device_node after snd_soc_register_card sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 5 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 6 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 7 +- .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 53 +++++---------- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 64 ++++++------------- 8 files changed, 56 insertions(+), 95 deletions(-) -- 2.34.1.448.ga2b2bfdf31-goog

2 5

[PATCH] ASoC: amd: acp: Power on/off the speaker enable gpio pin based on DAPM callback.
by V sujith kumar Reddy 24 Dec '21

24 Dec '21

Configure the speaker gpio pin based on power sequence of the DAPM speaker events. Enable speaker after widget power up and Disable before widget powerdown. Signed-off-by: V sujith kumar Reddy <vsujithkumar.reddy(a)amd.com> --- sound/soc/amd/acp/Kconfig | 6 +++--- sound/soc/amd/acp/acp-legacy-mach.c | 19 ++++++++++++++++--- sound/soc/amd/acp/acp-mach-common.c | 25 +++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 10 +++++++++- sound/soc/amd/acp/acp-sof-mach.c | 21 ++++++++++++++++++--- 5 files changed, 71 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 154be5e70821..d5838df3064b 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -32,7 +32,7 @@ config SND_AMD_ASOC_RENOIR config SND_SOC_AMD_MACH_COMMON tristate - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select CLK_FIXED_FCH select SND_SOC_RT5682_I2C select SND_SOC_DMIC @@ -44,14 +44,14 @@ config SND_SOC_AMD_MACH_COMMON config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select SND_SOC_AMD_MACH_COMMON help This option enables legacy sound card support for ACP audio. config SND_SOC_AMD_SOF_MACH tristate "AMD SOF Machine Driver Support" - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select SND_SOC_AMD_MACH_COMMON help This option enables SOF sound card support for ACP audio. diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index de0f8024e2fb..0ad1cf41b308 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -27,6 +27,7 @@ static struct acp_card_drvdata rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = NONE, + .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -41,15 +42,16 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), }; static int acp_asoc_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; + unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -67,9 +69,20 @@ static int acp_asoc_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_legacy_dai_links_create(card); + if (gpio_is_valid(spkr_gpio)) { + ret = devm_gpio_request(dev, spkr_gpio, "spkren"); + if (ret) { + dev_err(dev, "(%s) gpio request failed: %d\n", + __func__, ret); + return ret; + } + gpio_direction_output(spkr_gpio, 0); + } + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 7785f12aa006..03d8d1af14b3 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -71,6 +71,31 @@ static const struct snd_soc_dapm_route rt5682_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +int event_spkr_handler(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct acp_card_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(drvdata->gpio_spkr_en)) + return 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + gpio_set_value(drvdata->gpio_spkr_en, 1); + break; + case SND_SOC_DAPM_PRE_PMD: + gpio_set_value(drvdata->gpio_spkr_en, 0); + break; + default: + dev_warn(card->dev, "%s invalid setting\n", __func__); + break; + } + return 0; +} +EXPORT_SYMBOL_NS_GPL(event_spkr_handler, SND_SOC_AMD_MACH); + /* Define card ops for RT5682 CODEC */ static int acp_card_rt5682_init(struct snd_soc_pcm_runtime *rtd) { diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index 5dc47cfbff10..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -17,6 +17,12 @@ #include <linux/input.h> #include <linux/module.h> #include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> + +#define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 +#define EN_SPKR_GPIO_NONE -EINVAL enum be_id { HEADSET_BE_ID = 0, @@ -49,9 +55,11 @@ struct acp_card_drvdata { unsigned int dai_fmt; struct clk *wclk; struct clk *bclk; + unsigned int gpio_spkr_en; }; int acp_sofdsp_dai_links_create(struct snd_soc_card *card); int acp_legacy_dai_links_create(struct snd_soc_card *card); - +int event_spkr_handler(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event); #endif diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 854eb7214cea..07de46142655 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -27,6 +27,7 @@ static struct acp_card_drvdata sof_rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static struct acp_card_drvdata sof_rt5682_max_data = { @@ -36,6 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_max_data = { @@ -45,6 +47,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -58,15 +61,16 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), }; static int acp_sof_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; + unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -84,9 +88,20 @@ static int acp_sof_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_sofdsp_dai_links_create(card); + if (gpio_is_valid(spkr_gpio)) { + ret = devm_gpio_request(dev, spkr_gpio, "spkren"); + if (ret) { + dev_err(dev, "(%s) gpio request failed: %d\n", + __func__, ret); + return ret; + } + gpio_direction_output(spkr_gpio, 0); + } + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, -- 2.25.1

2 1

[PATCH] sound:soc:remove unneeded variable
by cgel.zte＠gmail.com 24 Dec '21

24 Dec '21

From: chiminghao <chi.minghao(a)zte.com.cn> return value form directly instead of taking this in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cm> Signed-off-by: chiminghao <chi.minghao(a)zte.com.cn> --- sound/soc/codecs/mt6660.c | 5 ++--- sound/soc/codecs/wcd938x.c | 6 ++---- 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index 358c500377df..3a881523c30f 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -47,13 +47,12 @@ static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) struct mt6660_chip *chip = context; int size = mt6660_get_reg_size(reg); u8 reg_data[4]; - int i, ret; + int i; for (i = 0; i < size; i++) reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; - ret = i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); - return ret; + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); } static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 52de7d14b139..0033d61d42e8 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3083,7 +3083,7 @@ static int wcd938x_mbhc_micb_ctrl_threshold_mic(struct snd_soc_component *compon int micb_num, bool req_en) { struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); - int rc, micb_mv; + int micb_mv; if (micb_num != MIC_BIAS_2) return -EINVAL; @@ -3097,9 +3097,7 @@ static int wcd938x_mbhc_micb_ctrl_threshold_mic(struct snd_soc_component *compon micb_mv = req_en ? WCD_MBHC_THR_HS_MICB_MV : wcd938x->micb2_mv; - rc = wcd938x_mbhc_micb_adjust_voltage(component, micb_mv, MIC_BIAS_2); - - return rc; + return wcd938x_mbhc_micb_adjust_voltage(component, micb_mv, MIC_BIAS_2); } static inline void wcd938x_mbhc_get_result_params(struct wcd938x_priv *wcd938x, -- 2.25.1

2 2

"alsatplg -v 1 -o -" silently corrupts output
by GitHub issues - opened 24 Dec '21

24 Dec '21

alsa-project/alsa-utils issue #131 was opened from marc-hb: `alsatplg -v 1` works. `alsatplg -o -` works too. Using both produces silently a binary .tplg file prefixed with debug logs: ``` B0x003d3b/015675 - pcm 'DMIC': write 912 bytes B0x0040cb/016587 - pcm 'HDMI1': write 912 bytes B0x00445b/017499 - pcm 'HDMI2': write 912 bytes B0x0047eb/018411 - pcm 'HDMI3': write 912 bytes B0x004b7b/019323 - pcm 'DMIC16kHz': write 912 bytes B0x004f0b/020235 - block size for type be (11:10) is 0x2ad4/10964 B0x004f0b/020235 - header index 0CoSA^E^@^@^@^@^@^@^@^H^@^@^@$^@^@ ^@^@^@^@^@s^@^@^@^@^@^@^@^A^@^@^@p^@^@^@ ^@^@^@.^@^@^@ ^@^@^@^F^@^ @^@^F^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^ @^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^ @^@^@^@^@^@^@^@^@^... ``` I would have expected alsatplg -v to use stderr, this would have avoided this problem. Probably too late by now. I suggest simply failing with an error message when both options are used at the same time. Any other solution that does not silently produces corrupted output would be a huge improvement. Context: thesofproject/sof/pull/5132 Issue URL : https://github.com/alsa-project/alsa-utils/issues/131 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RFC PATCH] ASoC: mediatek: mt8192-mt6359: fix device_node leak
by Tzung-Bi Shih 24 Dec '21

24 Dec '21

The of_parse_phandle() document: >>> Use of_node_put() on it when done. The driver didn't call of_node_put(). Fixes the leak. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Hi maintainers, Need your comments. The patch is one of the partial fixes that I think it makes the most sense. Option 1. Machine driver makes sure the object is valid until registered This patch adopts the option. It needs snd_soc_register_card() to call of_node_get() somewhere to hold the reference count of of_node. However, I failed to find similar logic in soc-core.c. Option 2. Machine driver borrows the reference count This is what [1] adopts. Decreasing the reference count in device's remove() to make sure the object is valid for whole sound card's lifecycle. [1]: https://elixir.bootlin.com/linux/v5.16-rc5/source/sound/soc/mediatek/mt8195… In fact, in my test environment, CONFIG_OF_DYNAMIC is not set. The implementation is empty[2]. Per [3]: >>> Hardly any platforms need this [2]: https://elixir.bootlin.com/linux/v5.16-rc5/source/include/linux/of.h#L125 [3]: https://elixir.bootlin.com/linux/v5.16-rc5/source/drivers/of/Kconfig#L55 I am not sure if it is worthy to find somewhere to hold the reference count in soc-core.c but I think option 1 makes it more clear to drivers. sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index a606133951b7..24a5d0adec1b 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1172,7 +1172,11 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) return ret; } - return devm_snd_soc_register_card(&pdev->dev, card); + ret = devm_snd_soc_register_card(&pdev->dev, card); + + of_node_put(platform_node); + of_node_put(hdmi_codec); + return ret; } #ifdef CONFIG_OF -- 2.34.1.173.g76aa8bc2d0-goog

2 2

[PATCH 0/4] ASoC: qcom: Parse "pin-switches" and "widgets" from DT
by Stephan Gerhold 23 Dec '21

23 Dec '21

Some sound card setups might require extra pin switches to allow turning off certain audio components. simple-card supports this already using the "pin-switches" and "widgets" device tree property. This series makes it possible to use the same properties for the Qcom sound cards. To implement that, the function that parses the "pin-switches" property in simple-card-utils.c is first moved into the ASoC core. Then two simple function calls are added to the common Qcom sound card DT parser. Finally there is a small patch for the msm8916-wcd-analog codec to make it possible to model sound card setups used in some MSM8916 smartphones. (See PATCH 2/4 for an explanation of some real example use cases.) Using pin switches rather than patching codec drivers with switches was originally suggested by Mark Brown on a patch for the tfa989x codec: https://lore.kernel.org/alsa-devel/YXaMVHo9drCIuD3u@sirena.org.uk/ Stephan Gerhold (4): ASoC: core: Add snd_soc_of_parse_pin_switches() from simple-card-utils ASoC: dt-bindings: qcom: sm8250: Document "pin-switches" and "widgets" ASoC: qcom: common: Parse "pin-switches" and "widgets" from DT ASoC: msm8916-wcd-analog: Use separate outputs for HPH_L/HPH_R .../bindings/sound/qcom,sm8250.yaml | 16 ++++++ include/sound/soc.h | 1 + sound/soc/codecs/msm8916-wcd-analog.c | 7 +-- sound/soc/generic/simple-card-utils.c | 45 +---------------- sound/soc/qcom/common.c | 10 ++++ sound/soc/soc-core.c | 50 +++++++++++++++++++ 6 files changed, 82 insertions(+), 47 deletions(-) -- 2.34.1

3 6

[PATCH 00/20] ASoC: SOF: Re-visit firmware state and panic tracking/handling
by Peter Ujfalusi 23 Dec '21

23 Dec '21

Hi, this series will improve how we are tracking the firmware's state to be able to avoid communication with it when it is not going to answer due to a panic and we will attempt to force power cycle the DSP to recover at the next runtime suspend time. The state handling brings in other improvements on the way the kernel reports errors and DSP panics to reduce the printed lines for normal users, but at the same time allowing developers (or for bug reports) to have more precise information available to track down the issue. We can now place messages easily in the correct debug level and not bound to the static ERROR for some of the print chains, causing excess amount or partial information to be printed, confusing users and machines (CI). I would have prefered to split this series up, but it was developed together to achieve a single goal to reduce the noise, but also provide the details we need to be able to rootcause issues. Regards, Peter --- Peter Ujfalusi (20): ASoC: SOF: ops: Use dev_warn() if the panic offsets differ ASoC: SOF: Intel: hda-loader: Avoid re-defining the HDA_FW_BOOT_ATTEMPTS ASoC: SOF: core: Add simple wrapper to check flags in sof_core_debug ASoC: SOF: Use sof_debug_check_flag() instead of sof_core_debug directly ASoC: SOF: Add 'non_recoverable' parameter to snd_sof_dsp_panic() ASoC: SOF: Add a 'message' parameter to snd_sof_dsp_dbg_dump() ASoC: SOF: Introduce new firmware state: SOF_FW_CRASHED ASoC: SOF: Introduce new firmware state: SOF_FW_BOOT_READY_OK ASoC: SOF: Move the definition of enum snd_sof_fw_state to global header ASoC: SOF: Rename 'enum snd_sof_fw_state' to 'enum sof_fw_state' ASoC: SOF: ipc: Only allow sending of an IPC in SOF_FW_BOOT_COMPLETE state ASoC: SOF: Set SOF_FW_BOOT_FAILED in case we have failure during boot ASoC: SOF: pm: Force DSP off on suspend in BOOT_FAILED state also ASoc: SOF: core: Update the FW boot state transition diagram ASoC: SOF: ops: Always print DSP Panic message but use different message ASoC: SOF: dsp_arch_ops: add kernel log level parameter for oops and stack ASoC: SOF: Rename snd_sof_get_status() and add kernel log level parameter ASoC: SOF: Add clarifying comments for sof_core_debug and DSP dump flags ASoC: SOF: debug: Use DEBUG log level for optional prints ASoC: SOF: Intel: hda: Use DEBUG log level for optional prints include/sound/sof.h | 22 ++++++ sound/soc/sof/core.c | 119 +++++++++++++++++++------------ sound/soc/sof/debug.c | 35 +++++---- sound/soc/sof/imx/imx-common.c | 4 +- sound/soc/sof/imx/imx8.c | 2 +- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/intel/atom.c | 8 +-- sound/soc/sof/intel/bdw.c | 8 +-- sound/soc/sof/intel/cnl.c | 21 +++++- sound/soc/sof/intel/hda-ipc.c | 19 ++++- sound/soc/sof/intel/hda-loader.c | 24 ++++--- sound/soc/sof/intel/hda.c | 20 +++--- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/ipc.c | 4 +- sound/soc/sof/loader.c | 16 ++--- sound/soc/sof/ops.c | 47 ++++++++---- sound/soc/sof/ops.h | 4 +- sound/soc/sof/pm.c | 10 +++ sound/soc/sof/sof-priv.h | 44 +++++------- sound/soc/sof/topology.c | 12 ++-- sound/soc/sof/xtensa/core.c | 44 +++++++----- 21 files changed, 299 insertions(+), 168 deletions(-) -- 2.34.1

2 21

[PATCH -next] ASoC: codec: tlv320adc3xxx: Fix missing clk_disable_unprepare() on error in adc3xxx_i2c_probe()
by Yang Yingliang 23 Dec '21

23 Dec '21

Fix the missing clk_disable_unprepare() before return from adc3xxx_i2c_probe() in the error handling case. Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/tlv320adc3xxx.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index a683bda7eb36..4baf3d881633 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -1232,21 +1232,21 @@ static int adc3xxx_i2c_probe(struct i2c_client *i2c, ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio_cfg[0]); if (ret < 0) - return ret; + goto err_unprepare_mclk; ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio_cfg[1]); if (ret < 0) - return ret; + goto err_unprepare_mclk; ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias1-vg", &adc3xxx->micbias_vg[0]); if (ret < 0) - return ret; + goto err_unprepare_mclk; ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias2-vg", &adc3xxx->micbias_vg[1]); if (ret < 0) - return ret; + goto err_unprepare_mclk; adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); if (IS_ERR(adc3xxx->regmap)) { ret = PTR_ERR(adc3xxx->regmap); - return ret; + goto err_unprepare_mclk; } i2c_set_clientdata(i2c, adc3xxx); @@ -1263,9 +1263,15 @@ static int adc3xxx_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_component(dev, &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); - if (ret < 0) + if (ret < 0) { dev_err(dev, "Failed to register codec: %d\n", ret); + goto err_unprepare_mclk; + } + + return 0; +err_unprepare_mclk: + clk_disable_unprepare(adc3xxx->mclk); return ret; } -- 2.25.1

2 1

[PATCH 0/5] ASoC: Intel: machine driver updates for 5.16
by Pierre-Louis Bossart 23 Dec '21

23 Dec '21

One quirk for a Dell TigerLake/SoundWire device, and initial support for platforms based on the ES8336 codec (aka ES8316). For full functionality, an update of the codec driver will be needed. Pierre-Louis Bossart (5): ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec ASoC: Intel: add machine driver for SOF+ES8336 ASoC: Intel: soc-acpi: add missing quirk for TGL SDCA single amp ASoC: Intel: sof_sdw: add missing quirk for Dell SKU 0A45 sound/hda/intel-dsp-config.c | 22 +- sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_es8336.c | 569 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw.c | 10 + .../intel/common/soc-acpi-intel-bxt-match.c | 6 + .../intel/common/soc-acpi-intel-glk-match.c | 7 +- .../intel/common/soc-acpi-intel-tgl-match.c | 47 ++ 8 files changed, 674 insertions(+), 3 deletions(-) create mode 100644 sound/soc/intel/boards/sof_es8336.c -- 2.25.1

3 10

[PATCH v3 0/3] Fix Tegra194 HDA regression
by Sameer Pujar 23 Dec '21

23 Dec '21

HDA probe failure is observed on Tegra194 based platforms and this happens due to reset failure. This series fixes the problem by skipping the failing reset and DT bindings are updated accordingly. Changelog ========= v2 -> v3: --------- * Use reset bulk APIs in HDA driver as suggested by Dmitry. v1 -> v2: --------- * Updated HDA driver patch to skip the failing reset instead of skipping resets in general for BPMP devices as per comment from Dmitry. * Used a better strucure name for SoC data as per comment from Thierry. * Dropped 'Fixes' tag in binding doc patch as per comment from Dmitry. Sameer Pujar (3): ALSA: hda/tegra: Fix Tegra194 HDA reset failure dt-bindings: sound: tegra: Update HDA resets arm64: tegra: Remove non existent Tegra194 reset .../bindings/sound/nvidia,tegra30-hda.yaml | 13 +++++-- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 5 +-- sound/pci/hda/hda_tegra.c | 45 +++++++++++++++++----- 3 files changed, 47 insertions(+), 16 deletions(-) -- 2.7.4

4 8

[GIT PULL] ASoC fixes for v5.16-rc6
by Mark Brown 22 Dec '21

22 Dec '21

The following changes since commit 3fc27e9a1f619b50700f020e6cd270c1b74755f0: ASoC: codecs: wsa881x: fix return values from kcontrol put (2021-12-01 14:13:53 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.16-rc6 for you to fetch changes up to ee907afb0c39a41ee74b862882cfe12820c74b98: ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s (2021-12-14 17:15:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.16 This is a relatively large set of driver specific changes so it may make sense to hold off to v5.17, though picking some over might be good. It's a combination of new device IDs and fixes for various driver specific things which are all small and of the usual "really bad if you're running into them" level, especially the Tegra ones. ---------------------------------------------------------------- Derek Fang (1): ASoC: rt5682: fix the wrong jack type detected Dmitry Osipenko (2): ASoC: tegra: Add DAPM switches for headphones and mic jack ASoC: tegra: Restore headphones jack name on Nyan Big Kai Vehmanen (2): ASoC: SOF: Intel: pci-tgl: add ADL-N support ASoC: SOF: Intel: pci-tgl: add new ADL-P variant Martin Blumenstingl (2): ASoC: meson: aiu: fifo: Add missing dma_coerce_mask_and_coherent() ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s Martin Povišer (1): ASoC: tas2770: Fix setting of high sample rates sound/soc/codecs/rt5682.c | 4 ++++ sound/soc/codecs/tas2770.c | 4 ++-- sound/soc/meson/aiu-encoder-i2s.c | 33 --------------------------------- sound/soc/meson/aiu-fifo-i2s.c | 19 +++++++++++++++++++ sound/soc/meson/aiu-fifo.c | 6 ++++++ sound/soc/sof/intel/pci-tgl.c | 4 ++++ sound/soc/tegra/tegra_asoc_machine.c | 11 ++++++++++- sound/soc/tegra/tegra_asoc_machine.h | 1 + 8 files changed, 46 insertions(+), 36 deletions(-)

2 1

[PATCH 0/2] ALSA: hda: intel-sdw-acpi: fix controller detection
by Pierre-Louis Bossart 22 Dec '21

22 Dec '21

The SoundWire-related information provided by platform firmware (DSDT) hasn't changed since 2016, but with a recent change of device hierarchy we need to change how the controller is detected. These patches were tested on existing devices and don't break anything, but are very much needed for newer devices (quirks already in Linus' branch for 5.16, see e.g. commit f55af7055cd4 ("ASoC: Intel: sof_sdw: Add support for SKU 0B12 product"). Unfortunately the BIOS changes happened *after* the initial patches were contributed. Libin Yang (2): ALSA: hda: intel-sdw-acpi: harden detection of controller ALSA: hda: intel-sdw-acpi: go through HDAS ACPI at max depth of 2 sound/hda/intel-sdw-acpi.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) -- 2.25.1

2 3

[PATCH 0/2] ALSA: hda: intel-sdw-acpi: go through HDAS ACPI at max depth of 2
by Bard Liao 22 Dec '21

22 Dec '21

In the HDAS ACPI scope, the SoundWire may not be the direct child of HDAS. It needs to go through the ACPI table at max depth of 2 to find the SoundWire device from HDAS. Libin Yang (2): ALSA: hda: intel-sdw-acpi: harden detection of controller ALSA: hda: intel-sdw-acpi: go through HDAS ACPI at max depth of 2 sound/hda/intel-sdw-acpi.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) -- 2.17.1

2 3

[PATCH] ALSA: rawmidi - fix the uninitalized user_pversion
by Jaroslav Kysela 22 Dec '21

22 Dec '21

The user_pversion was unitialized for the user space file structure in the open function, because the file private structure use kmalloc for the allocation. The kernel ALSA sequencer code clears the file structure, so no additional fixes are required. Cc: stable(a)kernel.org Cc: broonie(a)kernel.org BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Fixes: 09d23174402d ("ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION") Reported-by: syzbot+88412ee8811832b00dbe(a)syzkaller.appspotmail.com Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/rawmidi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6f30231bdb88..befa9809ff00 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -447,6 +447,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) err = -ENOMEM; goto __error; } + rawmidi_file->user_pversion = 0; init_waitqueue_entry(&wait, current); add_wait_queue(&rmidi->open_wait, &wait); while (1) { -- 2.31.1

2 1

[PATCH] ALSA: hda/realtek: Fix quirk for Clevo NJ51CU
by Werner Sembach 22 Dec '21

22 Dec '21

The Clevo NJ51CU comes either with the ALC293 or the ALC256 codec, but uses the 0x8686 subproduct id in both cases. The ALC256 codec needs a different quirk for the headset microphone working and and edditional quirk for sound working after suspend and resume. When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of 0x0020 on the ALC256 codec. Setting the value manually makes the sound work again. This patch does this automatically. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Fixes: b5acfe152abaa ("ALSA: hda/realtek: Add some Clove SSID in the ALC293(ALC1220)") Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3599f4c85ebf..3394ef209772 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6546,6 +6546,23 @@ static void alc233_fixup_no_audio_jack(struct hda_codec *codec, alc_process_coef_fw(codec, alc233_fixup_no_audio_jack_coefs); } +static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * The Clevo NJ51CU comes either with the ALC293 or the ALC256 codec, + * but uses the 0x8686 subproduct id in both cases. The ALC256 codec + * needs an additional quirk for sound working after suspend and resume. + */ + if (codec->core.vendor_id == 0x10ec0256) { + alc_update_coef_idx(codec, 0x10, 1<<9, 0); + snd_hda_codec_set_pincfg(codec, 0x19, 0x04a11120); + } + else + snd_hda_codec_set_pincfg(codec, 0x1a, 0x04a1113c); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6766,6 +6783,7 @@ enum { ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC233_FIXUP_NO_AUDIO_JACK, + ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME, }; static const struct hda_fixup alc269_fixups[] = { @@ -8490,6 +8508,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_no_audio_jack, }, + [ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_mic_no_presence_and_resume, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8829,7 +8853,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), SND_PCI_QUIRK(0x1558, 0x8a20, "Clevo NH55DCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8a51, "Clevo NH70RCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8d50, "Clevo NH55RCQ-M", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 1

Re: [PATCH 0/4] driver_core: Auxiliary drvdata helper cleanup
by Jason Gunthorpe 22 Dec '21

22 Dec '21

On Tue, Dec 21, 2021 at 04:48:17PM -0800, David E. Box wrote: > On Tue, 2021-12-21 at 20:09 -0400, Jason Gunthorpe wrote: > > On Tue, Dec 21, 2021 at 03:58:48PM -0800, David E. Box wrote: > > > Depends on "driver core: auxiliary bus: Add driver data helpers" patch [1]. > > > Applies the helpers to all auxiliary device drivers using > > > dev_(get/set)_drvdata. Drivers were found using the following search: > > > > > > grep -lr "struct auxiliary_device" $(grep -lr "drvdata" .) > > > > > > Changes were build tested using the following configs: > > > > > > vdpa/mlx5: CONFIG_MLX5_VDPA_NET > > > net/mlx53: CONFIG_MLX5_CORE_EN > > > soundwire/intel: CONFIG_SOUNDWIRE_INTEL > > > RDAM/irdma: CONFIG_INFINIBAND_IRDMA > > > CONFIG_MLX5_INFINIBAND > > > > > > [1] https://www.spinics.net/lists/platform-driver-x86/msg29940.html > > > > I have to say I don't really find this to be a big readability > > improvement. > > I should have referenced the thread [1] discussing the benefit of this change > since the question was asked and answered already. The idea is that drivers > shouldn't have to touch the device API directly if they are already using a > higher level core API (auxiliary bus) that can do that on its behalf. Driver writers should rarely use the auxilary device type directly, the should always immediately container_of it to their proper derived type. > > Also, what use is 'to_auxiliary_dev()' ? I didn't see any users added.. > > This was not added by that patch. It was added by the referenced patch, and seems totally pointless cut and paste, again because nothing should be using the auxiliary_device type for anything more than container_of'ing to their own type. We've been ripping out bus specific APIs in favour of generic ones (see the work on the DMA API for instance) so this whole concept seems regressive, particularly when applied to auxiliary bus which does not have an API of its own. Jason

1 0

[PATCH v2] dt-bindings: dsp: mediatek: add mt8195 dsp document
by YC Hung 22 Dec '21

22 Dec '21

From: "yc.hung" <yc.hung(a)mediatek.com> This patch adds mt8195 dsp document. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- Changes since v1: Rename yaml file name as mediatek,mt8195-dsp.yaml Refine descriptions for mailbox, memory-region and drop unused labels in examples. --- .../bindings/dsp/mediatek,mt8195-dsp.yaml | 117 ++++++++++++++++++ 1 file changed, 117 insertions(+) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml diff --git a/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml new file mode 100644 index 000000000000..bde763191d86 --- /dev/null +++ b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml @@ -0,0 +1,117 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/dsp/mediatek,mt8195-dsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek mt8195 DSP core + +maintainers: + - YC Hung <yc.hung(a)mediatek.com> + +description: | + Some boards from mt8195 contain a DSP core used for + advanced pre- and post- audio processing. + +properties: + compatible: + const: mediatek,mt8195-dsp + + reg: + items: + - description: Address and size of the DSP Cfg registers + - description: Address and size of the DSP SRAM + + reg-names: + items: + - const: cfg + - const: sram + + interrupts: + items: + - description: watchdog interrupt + + interrupt-names: + items: + - const: wdt + + clocks: + items: + - description: mux for audio dsp clock + - description: 26M clock + - description: mux for audio dsp local bus + - description: default audio dsp local bus clock source + - description: clock gate for audio dsp clock + - description: mux for audio dsp access external bus + + clock-names: + items: + - const: adsp_sel + - const: clk26m_ck + - const: audio_local_bus + - const: mainpll_d7_d2 + - const: scp_adsp_audiodsp + - const: audio_h + + power-domains: + maxItems: 1 + + mboxes: + items: + - description: ipc reply between host and audio DSP. + - description: ipc reuqest between host and audio DSP. + + mbox-names: + items: + - const: mbox0 + - const: mbox1 + + memory-region: + items: + - description: dma buffer between host and DSP. + - description: DSP system memory. + +required: + - compatible + - reg + - reg-names + - interrupts + - interrupt-names + - clocks + - clock-names + - memory-region + - power-domains + - mbox-names + - mboxes + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + dsp@10803000 { + compatible = "mediatek,mt8195-dsp"; + reg = <0x10803000 0x1000>, + <0x10840000 0x40000>; + reg-names = "cfg", "sram"; + interrupts = <GIC_SPI 694 IRQ_TYPE_LEVEL_HIGH 0>; + interrupt-names = "wdt"; + clocks = <&topckgen 10>, //CLK_TOP_ADSP + <&clk26m>, + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 136>, //CLK_TOP_MAINPLL_D7_D2 + <&scp_adsp 0>, //CLK_SCP_ADSP_AUDIODSP + <&topckgen 34>; //CLK_TOP_AUDIO_H + clock-names = "adsp_sel", + "clk26m_ck", + "audio_local_bus", + "mainpll_d7_d2", + "scp_adsp_audiodsp", + "audio_h"; + memory-region = <&adsp_dma_mem_reserved>, + <&adsp_mem_reserved>; + power-domains = <&spm 6>; //MT8195_POWER_DOMAIN_ADSP + mbox-names = "mbox0", "mbox1"; + mboxes = <&adsp_mailbox0>, <&adsp_mailbox1>; + }; -- 2.18.0

3 2

[PATCH] ASoC: codecs: ak4375: Change invert controls to a stereo switch
by Vincent Knecht 22 Dec '21

22 Dec '21

Don't use enums for DACL/DACR Signal Invert controls, and change them into a stereo "DAC Signal Invert Switch" control. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- sound/soc/codecs/ak4375.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/ak4375.c b/sound/soc/codecs/ak4375.c index a893aff42a01..22cda0699341 100644 --- a/sound/soc/codecs/ak4375.c +++ b/sound/soc/codecs/ak4375.c @@ -127,7 +127,6 @@ static DECLARE_TLV_DB_SCALE(hpg_tlv, -4200, 20, 0); static const char * const ak4375_ovolcn_select_texts[] = { "Dependent", "Independent" }; static const char * const ak4375_mdac_select_texts[] = { "x1", "x1/2" }; -static const char * const ak4375_inv_select_texts[] = { "Normal", "Inverting" }; static const char * const ak4375_cpmode_select_texts[] = { "Automatic Switching", "+-VDD Operation", @@ -157,12 +156,6 @@ static const struct soc_enum ak4375_mdacl_enum = static const struct soc_enum ak4375_mdacr_enum = SOC_ENUM_SINGLE(AK4375_07_DAC_MONO_MIXING, 6, ARRAY_SIZE(ak4375_mdac_select_texts), ak4375_mdac_select_texts); -static const struct soc_enum ak4375_invl_enum = - SOC_ENUM_SINGLE(AK4375_07_DAC_MONO_MIXING, 3, - ARRAY_SIZE(ak4375_inv_select_texts), ak4375_inv_select_texts); -static const struct soc_enum ak4375_invr_enum = - SOC_ENUM_SINGLE(AK4375_07_DAC_MONO_MIXING, 7, - ARRAY_SIZE(ak4375_inv_select_texts), ak4375_inv_select_texts); static const struct soc_enum ak4375_cpmode_enum = SOC_ENUM_SINGLE(AK4375_03_POWER_MANAGEMENT4, 2, ARRAY_SIZE(ak4375_cpmode_select_texts), ak4375_cpmode_select_texts); @@ -176,11 +169,11 @@ static const struct snd_kcontrol_new ak4375_snd_controls[] = { SOC_SINGLE_TLV("HP-Amp Analog Volume", AK4375_0D_HP_VOLUME_CONTROL, 0, 0x1f, 0, hpg_tlv), + SOC_DOUBLE("DAC Signal Invert Switch", AK4375_07_DAC_MONO_MIXING, 3, 7, 1, 0), + SOC_ENUM("Digital Volume Control", ak4375_ovolcn_enum), SOC_ENUM("DACL Signal Level", ak4375_mdacl_enum), SOC_ENUM("DACR Signal Level", ak4375_mdacr_enum), - SOC_ENUM("DACL Signal Invert", ak4375_invl_enum), - SOC_ENUM("DACR Signal Invert", ak4375_invr_enum), SOC_ENUM("Charge Pump Mode", ak4375_cpmode_enum), SOC_ENUM("DAC Digital Filter Mode", ak4375_digfil_enum), }; -- 2.31.1

2 1

[PATCH 1/2] ASoC: cs4265: Fix part number ID error message
by Fabio Estevam 22 Dec '21

22 Dec '21

The Chip ID - Register 01h contains the following description as per the CS4265 datasheet: "Bits 7 through 4 are the part number ID, which is 1101b (0Dh)" The current error message is incorrect as it prints CS4265_CHIP_ID, which is the register number, instead of printing the expected part number ID value. To make it clearer, also do a shift by 4, so that the error message would become: [ 4.218083] cs4265 1-004f: CS4265 Part Number ID: 0x0 Expected: 0xd Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index cffd6111afac..b89002189a2b 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -611,8 +611,8 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, if (devid != CS4265_CHIP_ID_VAL) { ret = -ENODEV; dev_err(&i2c_client->dev, - "CS4265 Device ID (%X). Expected %X\n", - devid, CS4265_CHIP_ID); + "CS4265 Part Number ID: 0x%x Expected: 0x%x\n", + devid >> 4, CS4265_CHIP_ID_VAL >> 4); return ret; } dev_info(&i2c_client->dev, -- 2.25.1

1 1

[PATCH v2 1/3] ASoC: dt-bindings: Use name-prefix schema
by Alexander Stein 22 Dec '21

22 Dec '21

name-prefix.txt does not exist anymore, just reference the schema instead. Signed-off-by: Alexander Stein <alexander.stein(a)mailbox.org> --- Changes in v2: * Fix 'pass' -> 'true' .../devicetree/bindings/sound/simple-audio-amplifier.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml index 26379377a7ac..8327846356d3 100644 --- a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -9,6 +9,9 @@ title: Simple Audio Amplifier Device Tree Bindings maintainers: - Jerome Brunet <jbrunet(a)baylibre.com> +allOf: + - $ref: name-prefix.yaml# + properties: compatible: enum: @@ -22,10 +25,7 @@ properties: description: > power supply for the device - sound-name-prefix: - $ref: /schemas/types.yaml#/definitions/string - description: > - See ./name-prefix.txt + sound-name-prefix: true required: - compatible -- 2.34.1

4 7

Re: [PATCH 0/4] driver_core: Auxiliary drvdata helper cleanup
by Cezary Rojewski 22 Dec '21

22 Dec '21

On 2021-12-22 12:58 AM, David E. Box wrote: > Depends on "driver core: auxiliary bus: Add driver data helpers" patch [1]. > Applies the helpers to all auxiliary device drivers using > dev_(get/set)_drvdata. Drivers were found using the following search: > > grep -lr "struct auxiliary_device" $(grep -lr "drvdata" .) > > Changes were build tested using the following configs: > > vdpa/mlx5: CONFIG_MLX5_VDPA_NET > net/mlx53: CONFIG_MLX5_CORE_EN > soundwire/intel: CONFIG_SOUNDWIRE_INTEL > RDAM/irdma: CONFIG_INFINIBAND_IRDMA > CONFIG_MLX5_INFINIBAND > > [1] https://www.spinics.net/lists/platform-driver-x86/msg29940.html > > David E. Box (4): > RDMA/irdma: Use auxiliary_device driver data helpers > soundwire: intel: Use auxiliary_device driver data helpers > net/mlx5e: Use auxiliary_device driver data helpers > vdpa/mlx5: Use auxiliary_device driver data helpers > > drivers/infiniband/hw/irdma/main.c | 4 ++-- > drivers/infiniband/hw/mlx5/main.c | 8 ++++---- > drivers/net/ethernet/mellanox/mlx5/core/en_main.c | 8 ++++---- > drivers/soundwire/intel.c | 8 ++++---- > drivers/soundwire/intel_init.c | 2 +- > drivers/vdpa/mlx5/net/mlx5_vnet.c | 4 ++-- > 6 files changed, 17 insertions(+), 17 deletions(-) Changes look good, I did post one question regarding possible occurrence of dev_get_drvdata() not being accounted for in patch: [PATCH 1/4] RDMA/irdma: Use auxiliary_device driver data helpers However, it does look like a blocker so: Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> For the record, I did scan the following files (users of struct auxiliary_device) from drivers/ directory for the missing occurrences and found only a single one (as mentioned earlier): drivers/net/ethernet/mellanox/mlx5/core/dev.c drivers/net/ethernet/mellanox/mlx5/core/sf/dev/dev.c drivers/net/ethernet/mellanox/mlx5/core/sf/dev/driver.c drivers/net/ethernet/mellanox/mlx5/core/sf/dev/dev.h drivers/net/ethernet/mellanox/mlx5/core/en_rep.c drivers/net/ethernet/mellanox/mlx5/core/en_main.c drivers/net/ethernet/intel/i40e/i40e_client.c drivers/net/ethernet/intel/ice/ice.h drivers/net/ethernet/intel/ice/ice_idc.c drivers/infiniband/hw/irdma/main.c drivers/infiniband/hw/irdma/i40iw_if.c drivers/infiniband/hw/mlx5/main.c drivers/infiniband/hw/mlx5/ib_rep.c drivers/vdpa/mlx5/net/mlx5_vnet.c drivers/gpu/drm/bridge/ti-sn65dsi86.c drivers/soundwire/intel_init.c drivers/soundwire/intel.c Regards, Czarek

1 0

Re: [PATCH 1/4] RDMA/irdma: Use auxiliary_device driver data helpers
by Cezary Rojewski 22 Dec '21

22 Dec '21

On 2021-12-22 12:58 AM, David E. Box wrote: > Use auxiliary_get_drvdata and auxiliary_set_drvdata helpers. > > Signed-off-by: David E. Box <david.e.box(a)linux.intel.com> > --- > drivers/infiniband/hw/irdma/main.c | 4 ++-- > drivers/infiniband/hw/mlx5/main.c | 8 ++++---- > 2 files changed, 6 insertions(+), 6 deletions(-) > > diff --git a/drivers/infiniband/hw/irdma/main.c b/drivers/infiniband/hw/irdma/main.c > index 51a41359e0b4..9ccf4d683f8a 100644 > --- a/drivers/infiniband/hw/irdma/main.c > +++ b/drivers/infiniband/hw/irdma/main.c While two occurrences of aux_dev->dev have been addressed here for irdma/main.c, there is one more that probably could get updated too: static void irdma_iidc_event_handler(struct ice_pf *pf, struct iidrc_event *event) { struct irdma_device *iwdev = dev_get_drvdata(&pf->adev->dev); (...) } Note: the declaration of struct ice_pf reads: struct ice_pf { (...) struct auxiliary_device *adev; (...) }; leads into suggestion: struct irdma_device *iwdev = auxiliary_get_drvdata(pf->adev); Of course, even if I'm right about this, such change could be applied with a separate patch and does not block the current review. Regards, Czarek

1 0

Re: [PATCH 2/4] soundwire: intel: Use auxiliary_device driver data helpers
by Andy Shevchenko 22 Dec '21

22 Dec '21

On Tue, Dec 21, 2021 at 03:58:50PM -0800, David E. Box wrote: > Use auxiliary_get_drvdata and auxiliary_set_drvdata helpers. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> > Signed-off-by: David E. Box <david.e.box(a)linux.intel.com> > --- > drivers/soundwire/intel.c | 8 ++++---- > drivers/soundwire/intel_init.c | 2 +- > 2 files changed, 5 insertions(+), 5 deletions(-) > > diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c > index 78037ffdb09b..d082d18e41a9 100644 > --- a/drivers/soundwire/intel.c > +++ b/drivers/soundwire/intel.c > @@ -1293,7 +1293,7 @@ static int intel_link_probe(struct auxiliary_device *auxdev, > bus->ops = &sdw_intel_ops; > > /* set driver data, accessed by snd_soc_dai_get_drvdata() */ > - dev_set_drvdata(dev, cdns); > + auxiliary_set_drvdata(auxdev, cdns); > > /* use generic bandwidth allocation algorithm */ > sdw->cdns.bus.compute_params = sdw_compute_params; > @@ -1321,7 +1321,7 @@ int intel_link_startup(struct auxiliary_device *auxdev) > { > struct sdw_cdns_stream_config config; > struct device *dev = &auxdev->dev; > - struct sdw_cdns *cdns = dev_get_drvdata(dev); > + struct sdw_cdns *cdns = auxiliary_get_drvdata(auxdev); > struct sdw_intel *sdw = cdns_to_intel(cdns); > struct sdw_bus *bus = &cdns->bus; > int link_flags; > @@ -1463,7 +1463,7 @@ int intel_link_startup(struct auxiliary_device *auxdev) > static void intel_link_remove(struct auxiliary_device *auxdev) > { > struct device *dev = &auxdev->dev; > - struct sdw_cdns *cdns = dev_get_drvdata(dev); > + struct sdw_cdns *cdns = auxiliary_get_drvdata(auxdev); > struct sdw_intel *sdw = cdns_to_intel(cdns); > struct sdw_bus *bus = &cdns->bus; > > @@ -1488,7 +1488,7 @@ int intel_link_process_wakeen_event(struct auxiliary_device *auxdev) > void __iomem *shim; > u16 wake_sts; > > - sdw = dev_get_drvdata(dev); > + sdw = auxiliary_get_drvdata(auxdev); > bus = &sdw->cdns.bus; > > if (bus->prop.hw_disabled || !sdw->startup_done) { > diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c > index e329022e1669..d99807765dfe 100644 > --- a/drivers/soundwire/intel_init.c > +++ b/drivers/soundwire/intel_init.c > @@ -244,7 +244,7 @@ static struct sdw_intel_ctx > goto err; > > link = &ldev->link_res; > - link->cdns = dev_get_drvdata(&ldev->auxdev.dev); > + link->cdns = auxiliary_get_drvdata(&ldev->auxdev); > > if (!link->cdns) { > dev_err(&adev->dev, "failed to get link->cdns\n"); > -- > 2.25.1 > -- With Best Regards, Andy Shevchenko

1 0

compile error: undefined reference to `snd_ctl_elem_id_compare_set'
by GitHub issues - opened 22 Dec '21

22 Dec '21

alsa-project/alsa-utils issue #130 was opened from steeley: Building on raspberry pi buster (Linux 5.10.76-v7l. 32 bit) using theses commands: `cd alsa-utils/ git checkout v1.2.6 autoreconf -i ./configure --prefix=/usr/local make -j4` I get this error: `gcc -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -g -O2 -o alsactl alsactl-alsactl.o alsactl-state.o alsactl-lock.o alsactl-utils.o alsactl-init_parse.o alsactl-init_ucm.o alsactl-daemon.o alsactl-monitor.o alsactl-clean.o -lasound -lm -ldl -lpthread /usr/bin/ld: alsactl-clean.o: in function `clean_one_control': /home/pi/alsa-utils/alsactl/clean.c:55: undefined reference to `snd_ctl_elem_id_compare_set' collect2: error: ld returned 1 exit status make[2]: *** [Makefile:528: alsactl] Error 1 make[2]: Leaving directory '/home/pi/alsa-utils/alsactl' make[1]: *** [Makefile:827: all-recursive] Error 1 make[1]: Leaving directory '/home/pi/alsa-utils/alsactl' make: *** [Makefile:429: all-recursive] Error 1` Issue URL : https://github.com/alsa-project/alsa-utils/issues/130 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3 1/1] pinctrl: Propagate firmware node from a parent device
by Andy Shevchenko 22 Dec '21

22 Dec '21

When creating MFD platform devices the firmware node is left unset. This, in particular, prevents GPIO library to use it for different purposes. Propagate firmware node from the parent device and let GPIO library do the right thing. While at it, slightly modify the headers to reflect the usage of APIs. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- v2: fixed more compilation errors brought by header reshuffling (LKP) drivers/pinctrl/cirrus/pinctrl-madera-core.c | 5 ++++- drivers/pinctrl/pinctrl-as3722.c | 13 +++++++------ drivers/pinctrl/pinctrl-da9062.c | 6 +++--- drivers/pinctrl/pinctrl-max77620.c | 11 +++++++---- drivers/pinctrl/pinctrl-rk805.c | 12 ++++++------ 5 files changed, 27 insertions(+), 20 deletions(-) diff --git a/drivers/pinctrl/cirrus/pinctrl-madera-core.c b/drivers/pinctrl/cirrus/pinctrl-madera-core.c index dce2626384a9..e1cfbee3643a 100644 --- a/drivers/pinctrl/cirrus/pinctrl-madera-core.c +++ b/drivers/pinctrl/cirrus/pinctrl-madera-core.c @@ -8,8 +8,10 @@ #include <linux/err.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/property.h> #include <linux/regmap.h> #include <linux/slab.h> + #include <linux/pinctrl/machine.h> #include <linux/pinctrl/pinctrl.h> #include <linux/pinctrl/pinmux.h> @@ -1004,13 +1006,14 @@ static int madera_pin_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s\n", __func__); + device_set_node(&pdev->dev, dev_fwnode(pdev->dev.parent)); + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; priv->dev = &pdev->dev; priv->madera = madera; - pdev->dev.of_node = madera->dev->of_node; switch (madera->type) { case CS47L15: diff --git a/drivers/pinctrl/pinctrl-as3722.c b/drivers/pinctrl/pinctrl-as3722.c index 13c193156363..4313756b52e6 100644 --- a/drivers/pinctrl/pinctrl-as3722.c +++ b/drivers/pinctrl/pinctrl-as3722.c @@ -23,19 +23,20 @@ #include <linux/delay.h> #include <linux/gpio/driver.h> #include <linux/kernel.h> +#include <linux/mod_devicetable.h> #include <linux/module.h> #include <linux/mfd/as3722.h> -#include <linux/of.h> -#include <linux/of_device.h> #include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/property.h> +#include <linux/slab.h> + #include <linux/pinctrl/consumer.h> #include <linux/pinctrl/machine.h> #include <linux/pinctrl/pinctrl.h> #include <linux/pinctrl/pinconf-generic.h> #include <linux/pinctrl/pinconf.h> #include <linux/pinctrl/pinmux.h> -#include <linux/pm.h> -#include <linux/slab.h> #include "core.h" #include "pinconf.h" @@ -551,12 +552,13 @@ static int as3722_pinctrl_probe(struct platform_device *pdev) struct as3722_pctrl_info *as_pci; int ret; + device_set_node(&pdev->dev, dev_fwnode(pdev->dev.parent)); + as_pci = devm_kzalloc(&pdev->dev, sizeof(*as_pci), GFP_KERNEL); if (!as_pci) return -ENOMEM; as_pci->dev = &pdev->dev; - as_pci->dev->of_node = pdev->dev.parent->of_node; as_pci->as3722 = dev_get_drvdata(pdev->dev.parent); platform_set_drvdata(pdev, as_pci); @@ -578,7 +580,6 @@ static int as3722_pinctrl_probe(struct platform_device *pdev) as_pci->gpio_chip = as3722_gpio_chip; as_pci->gpio_chip.parent = &pdev->dev; - as_pci->gpio_chip.of_node = pdev->dev.parent->of_node; ret = gpiochip_add_data(&as_pci->gpio_chip, as_pci); if (ret < 0) { dev_err(&pdev->dev, "Couldn't register gpiochip, %d\n", ret); diff --git a/drivers/pinctrl/pinctrl-da9062.c b/drivers/pinctrl/pinctrl-da9062.c index 1c08579f0198..0e0ac3f3ffef 100644 --- a/drivers/pinctrl/pinctrl-da9062.c +++ b/drivers/pinctrl/pinctrl-da9062.c @@ -14,6 +14,7 @@ #include <linux/bits.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/property.h> #include <linux/regmap.h> #include <linux/gpio/driver.h> @@ -256,6 +257,8 @@ static int da9062_pctl_probe(struct platform_device *pdev) struct da9062_pctl *pctl; int i; + device_set_node(&pdev->dev, dev_fwnode(pdev->dev.parent)); + pctl = devm_kzalloc(&pdev->dev, sizeof(*pctl), GFP_KERNEL); if (!pctl) return -ENOMEM; @@ -277,9 +280,6 @@ static int da9062_pctl_probe(struct platform_device *pdev) pctl->gc = reference_gc; pctl->gc.label = dev_name(&pdev->dev); pctl->gc.parent = &pdev->dev; -#ifdef CONFIG_OF_GPIO - pctl->gc.of_node = parent->of_node; -#endif platform_set_drvdata(pdev, pctl); diff --git a/drivers/pinctrl/pinctrl-max77620.c b/drivers/pinctrl/pinctrl-max77620.c index c643ed43ebbf..1ee94574f0af 100644 --- a/drivers/pinctrl/pinctrl-max77620.c +++ b/drivers/pinctrl/pinctrl-max77620.c @@ -10,14 +10,16 @@ */ #include <linux/mfd/max77620.h> +#include <linux/mod_devicetable.h> #include <linux/module.h> -#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/property.h> +#include <linux/regmap.h> + #include <linux/pinctrl/pinctrl.h> #include <linux/pinctrl/pinconf-generic.h> #include <linux/pinctrl/pinconf.h> #include <linux/pinctrl/pinmux.h> -#include <linux/platform_device.h> -#include <linux/regmap.h> #include "core.h" #include "pinconf.h" @@ -551,12 +553,13 @@ static int max77620_pinctrl_probe(struct platform_device *pdev) struct max77620_pctrl_info *mpci; int i; + device_set_node(&pdev->dev, dev_fwnode(pdev->dev.parent)); + mpci = devm_kzalloc(&pdev->dev, sizeof(*mpci), GFP_KERNEL); if (!mpci) return -ENOMEM; mpci->dev = &pdev->dev; - mpci->dev->of_node = pdev->dev.parent->of_node; mpci->rmap = max77620->rmap; mpci->pins = max77620_pins_desc; diff --git a/drivers/pinctrl/pinctrl-rk805.c b/drivers/pinctrl/pinctrl-rk805.c index c6f4229eb106..7c1f7408fb9a 100644 --- a/drivers/pinctrl/pinctrl-rk805.c +++ b/drivers/pinctrl/pinctrl-rk805.c @@ -13,17 +13,17 @@ #include <linux/kernel.h> #include <linux/module.h> #include <linux/mfd/rk808.h> -#include <linux/of.h> -#include <linux/of_device.h> #include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/property.h> +#include <linux/slab.h> + #include <linux/pinctrl/consumer.h> #include <linux/pinctrl/machine.h> #include <linux/pinctrl/pinctrl.h> #include <linux/pinctrl/pinconf-generic.h> #include <linux/pinctrl/pinconf.h> #include <linux/pinctrl/pinmux.h> -#include <linux/pm.h> -#include <linux/slab.h> #include "core.h" #include "pinconf.h" @@ -420,18 +420,18 @@ static int rk805_pinctrl_probe(struct platform_device *pdev) struct rk805_pctrl_info *pci; int ret; + device_set_node(&pdev->dev, dev_fwnode(pdev->dev.parent)); + pci = devm_kzalloc(&pdev->dev, sizeof(*pci), GFP_KERNEL); if (!pci) return -ENOMEM; pci->dev = &pdev->dev; - pci->dev->of_node = pdev->dev.parent->of_node; pci->rk808 = dev_get_drvdata(pdev->dev.parent); pci->pinctrl_desc = rk805_pinctrl_desc; pci->gpio_chip = rk805_gpio_chip; pci->gpio_chip.parent = &pdev->dev; - pci->gpio_chip.of_node = pdev->dev.parent->of_node; platform_set_drvdata(pdev, pci); -- 2.34.1

4 3

Re: [PATCH 0/4] driver_core: Auxiliary drvdata helper cleanup
by Jason Gunthorpe 22 Dec '21

22 Dec '21

On Tue, Dec 21, 2021 at 03:58:48PM -0800, David E. Box wrote: > Depends on "driver core: auxiliary bus: Add driver data helpers" patch [1]. > Applies the helpers to all auxiliary device drivers using > dev_(get/set)_drvdata. Drivers were found using the following search: > > grep -lr "struct auxiliary_device" $(grep -lr "drvdata" .) > > Changes were build tested using the following configs: > > vdpa/mlx5: CONFIG_MLX5_VDPA_NET > net/mlx53: CONFIG_MLX5_CORE_EN > soundwire/intel: CONFIG_SOUNDWIRE_INTEL > RDAM/irdma: CONFIG_INFINIBAND_IRDMA > CONFIG_MLX5_INFINIBAND > > [1] https://www.spinics.net/lists/platform-driver-x86/msg29940.html I have to say I don't really find this to be a big readability improvement. Also, what use is 'to_auxiliary_dev()' ? I didn't see any users added.. Jason

1 0

[WIP][RFC] Nhlt plugin and simple plugin interface
by GitHub pull_request - opened 21 Dec '21

21 Dec '21

alsa-project/alsa-utils pull request #129 was opened from juimonen: Initial something... I need to figure out: - where to specify which plugins to load/run (nhlt now hard coded) - where to install the plugins - how to get the to-be-loaded plugin names and where - is the "hidden-visibility" gcc flag ok to hide other than .so entrypoint Request URL : https://github.com/alsa-project/alsa-utils/pull/129 Patch URL : https://github.com/alsa-project/alsa-utils/pull/129.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 1/3] ASoC: SOF: AMD: simplify return status handling
by Ajit Kumar Pandey 21 Dec '21

21 Dec '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> cppcheck warning: sound/soc/sof/amd/acp.c:222:9: warning: Identical condition and return expression 'ret', return value is always 0 [identicalConditionAfterEarlyExit] return ret; ^ sound/soc/sof/amd/acp.c:213:6: note: If condition 'ret' is true, the function will return/exit if (ret) ^ sound/soc/sof/amd/acp.c:222:9: note: Returning identical expression 'ret' return ret; ^ Just return 0; on success. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/sof/amd/acp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 4c5550e8d364..fe9b7dc5bc86 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -219,7 +219,7 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return -EINVAL; } - return ret; + return 0; } int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) -- 2.25.1

2 3

[PATCH v2 1/2] ASoC: dt-bindings: codecs: Add bindings for ak4375
by Vincent Knecht 21 Dec '21

21 Dec '21

AK4375 is an audio DAC with headphones amplifier controlled via I2C. Add simple device tree bindings that describe how to set it up. Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- Changed in v2: - Changed avdd-supply to be required - Picked up Rob's Reviewed-by Sorry if it was not appropriate following the avdd change... --- .../devicetree/bindings/sound/ak4375.yaml | 57 +++++++++++++++++++ 1 file changed, 57 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4375.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4375.yaml b/Documentation/devicetree/bindings/sound/ak4375.yaml new file mode 100644 index 000000000000..f1d5074a024d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4375.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4375.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4375 DAC and headphones amplifier Device Tree Bindings + +maintainers: + - Vincent Knecht <vincent.knecht(a)mailoo.org> + +properties: + compatible: + const: asahi-kasei,ak4375 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + avdd-supply: + description: regulator phandle for the AVDD power supply. + + tvdd-supply: + description: regulator phandle for the TVDD power supply. + + pdn-gpios: + description: optional GPIO to set the PDN pin. + +required: + - compatible + - reg + - '#sound-dai-cells' + - avdd-supply + - tvdd-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + headphones: audio-codec@10 { + compatible = "asahi-kasei,ak4375"; + reg = <0x10>; + avdd-supply = <&reg_headphones_avdd>; + tvdd-supply = <&pm8916_l6>; + pdn-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&headphones_pdn_default>; + #sound-dai-cells = <0>; + }; + }; -- 2.31.1

2 3

[PATCH v2 0/2] sound/soc: Use platform_get_irq() to fetch IRQ's
by Lad Prabhakar 21 Dec '21

21 Dec '21

Hi All, This patch series aims to drop using platform_get_resource() for IRQ types in preparation for removal of static setup of IRQ resource from DT core code. Dropping usage of platform_get_resource() was agreed based on the discussion [0]. [0] https://patchwork.kernel.org/project/linux-renesas-soc/ patch/20211209001056.29774-1-prabhakar.mahadev-lad.rj(a)bp.renesas.com/ Cheers, Prabhakar Lad Prabhakar (2): ASoC: xlnx: Use platform_get_irq() to get the interrupt ASoC: bcm: Use platform_get_irq() to get the interrupt sound/soc/bcm/bcm63xx-i2s.h | 1 - sound/soc/bcm/bcm63xx-pcm-whistler.c | 13 ++++++------- sound/soc/xilinx/xlnx_spdif.c | 10 +++------- 3 files changed, 9 insertions(+), 15 deletions(-) -- 2.17.1

2 3

[PATCH] ASoC: amd: acp: Remove duplicate dependency in Kconfig
by Ajit Kumar Pandey 21 Dec '21

21 Dec '21

Remove duplicate depends on statement in Kconfig file. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/amd/acp/Kconfig | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 52a1371f9e61..154be5e70821 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -46,7 +46,6 @@ config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" depends on X86 && PCI && I2C select SND_SOC_AMD_MACH_COMMON - depends on X86 && PCI && I2C help This option enables legacy sound card support for ACP audio. @@ -54,7 +53,6 @@ config SND_SOC_AMD_SOF_MACH tristate "AMD SOF Machine Driver Support" depends on X86 && PCI && I2C select SND_SOC_AMD_MACH_COMMON - depends on X86 && PCI && I2C help This option enables SOF sound card support for ACP audio. -- 2.25.1

2 1

[PATCH 0/2] sound/soc: Use platform_get_irq() to fetch IRQ's
by Lad Prabhakar 21 Dec '21

21 Dec '21

Hi All, This patch series aims to drop using platform_get_resource() for IRQ types in preparation for removal of static setup of IRQ resource from DT core code. Dropping usage of platform_get_resource() was agreed based on the discussion [0]. [0] https://patchwork.kernel.org/project/linux-renesas-soc/ patch/20211209001056.29774-1-prabhakar.mahadev-lad.rj(a)bp.renesas.com/ Note: I have just build tested the patches. Cheers, Prabhakar Lad Prabhakar (2): ASoC: xlnx: Use platform_get_irq() to get the interrupt ASoC: bcm: Use platform_get_irq() to get the interrupt sound/soc/bcm/bcm63xx-i2s.h | 1 - sound/soc/bcm/bcm63xx-pcm-whistler.c | 12 +++++------- sound/soc/xilinx/xlnx_spdif.c | 10 +++------- 3 files changed, 8 insertions(+), 15 deletions(-) -- 2.17.1

3 5

[PATCH v1] dt-bindings: sound: tegra-audio-rt5677: Correct example
by Dmitry Osipenko 21 Dec '21

21 Dec '21

Remove non-existent properties from the example of the binding. These properties were borrower from the old txt binding, but they were never used in practice and aren't documented in the new binding. They aren't reported by the binding checker because dtschema needs extra patch that hasn't been upstreamed yet to make unevaluatedProperties work properly. Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- .../devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml | 3 --- 1 file changed, 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml index 03ff691c26c8..a49997d6028b 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml @@ -92,9 +92,6 @@ examples: nvidia,audio-codec = <&rt5677>; nvidia,hp-det-gpios = <&gpio 143 0>; - nvidia,mic-present-gpios = <&gpio 132 1>; - nvidia,hp-en-gpios = <&rt5677 1 0>; - nvidia,dmic-clk-en-gpios = <&rt5677 2 1>; clocks = <&clk 216>, <&clk 217>, -- 2.33.1

3 4

[PATCH 1/3] ASoC: SOF: AMD: simplify return status handling
by Ajit Kumar Pandey 21 Dec '21

21 Dec '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> cppcheck warning: sound/soc/sof/amd/acp.c:222:9: warning: Identical condition and return expression 'ret', return value is always 0 [identicalConditionAfterEarlyExit] return ret; ^ sound/soc/sof/amd/acp.c:213:6: note: If condition 'ret' is true, the function will return/exit if (ret) ^ sound/soc/sof/amd/acp.c:222:9: note: Returning identical expression 'ret' return ret; ^ Just return 0; on success. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/amd/acp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 4c5550e8d364..fe9b7dc5bc86 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -219,7 +219,7 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return -EINVAL; } - return ret; + return 0; } int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) -- 2.25.1

2 5

[PATCH v2 0/3] Fix Tegra194 HDA regression
by Sameer Pujar 21 Dec '21

21 Dec '21

HDA probe failure is observed on Tegra194 based platforms and this happens due to reset failure. This series fixes the problem by skipping the failing reset and DT bindings are updated accordingly. Changelog ========= v1 -> v2: --------- * Updated HDA driver patch to skip the failing reset instead of skipping resets in general for BPMP devices as per comment from Dmitry. * Used a better strucure name for SoC data as per comment from Thierry. * Dropped 'Fixes' tag in binding doc patch as per comment from Dmitry. Sameer Pujar (3): ALSA: hda/tegra: Fix Tegra194 HDA reset failure dt-bindings: sound: tegra: Update HDA resets arm64: tegra: Remove non existent Tegra194 reset .../bindings/sound/nvidia,tegra30-hda.yaml | 13 ++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 5 +- sound/pci/hda/hda_tegra.c | 96 +++++++++++++++++++--- 3 files changed, 97 insertions(+), 17 deletions(-) -- 2.7.4

2 7

[PATCH] ASoC: mediatek: mt8195-mt6359: reduce log verbosity in probe()
by Tzung-Bi Shih 21 Dec '21

21 Dec '21

Eliminates error messages if snd_soc_register_card() failed. Kernel emits messages if device probe error anyway. This is mainly for removing the following error messages during boot. >>> snd_soc_register_card fail -517 Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c | 2 -- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c index 5cdbfaafd479..9cf907c49ea8 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c @@ -1114,8 +1114,6 @@ static int mt8195_mt6359_rt1011_rt5682_dev_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", - __func__, ret); of_node_put(priv->hdmi_node); of_node_put(priv->dp_node); of_node_put(priv->platform_node); diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index fa50a31e9718..fdd444138728 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -1358,8 +1358,6 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", - __func__, ret); of_node_put(priv->hdmi_node); of_node_put(priv->dp_node); of_node_put(priv->platform_node); -- 2.34.1.173.g76aa8bc2d0-goog

2 1

[PATCH v7 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 21 Dec '21

21 Dec '21

DT bindings for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Change-Id: I94e948892f47df7acb5cffc007d4678b37a96bc8 Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..83936f594d1a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: audio-codec@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..ec988439da20 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[PATCH v7 2/2] ASoC: codec: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 21 Dec '21

21 Dec '21

New codec driver for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1311 ++++++++++++++++++++++++++++++ 3 files changed, 1321 insertions(+) create mode 100644 sound/soc/codecs/tlv320adc3xxx.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c033ee7d82e4..22836ca9b478 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -220,6 +220,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 imply SND_SOC_TFA989X + imply SND_SOC_TLV320ADC3XXX imply SND_SOC_TLV320ADCX140 imply SND_SOC_TLV320AIC23_I2C imply SND_SOC_TLV320AIC23_SPI @@ -1496,6 +1497,13 @@ config SND_SOC_TFA989X Note that the driver currently bypasses the built-in "CoolFlux DSP" and does not support (hardware) volume control. +config SND_SOC_TLV320ADC3XXX + tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" + depends on I2C + help + Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 + ADCs. + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 485eee75502b..24bc6b34ba2f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -239,6 +239,7 @@ snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o +snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o @@ -574,6 +575,7 @@ obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TFA989X) += snd-soc-tfa989x.o +obj-$(CONFIG_SND_SOC_TLV320ADC3XXX) += snd-soc-tlv320adc3xxx.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c new file mode 100644 index 000000000000..a683bda7eb36 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -0,0 +1,1311 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov +// +// Copyright (C) 2010 Mistral Solutions Pvt Ltd. +// Author: Shahina Shaik <shahina.s(a)mistralsolutions.com> +// +// Copyright (C) 2014-2018, Ambarella, Inc. +// Author: Dongge wu <dgwu(a)ambarella.com> +// +// Copyright (C) 2021 Axis Communications AB +// Author: Ricard Wanderlof <ricardw(a)axis.com> +// + +#include <dt-bindings/sound/tlv320adc3xxx.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio/driver.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +/* + * General definitions defining exported functionality. + */ + +#define ADC3XXX_MICBIAS_PINS 2 + +/* Number of GPIO pins exposed via the gpiolib interface */ +#define ADC3XXX_GPIOS_MAX 2 + +#define ADC3XXX_RATES SNDRV_PCM_RATE_8000_96000 +#define ADC3XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * PLL modes, to be used for clk_id for set_sysclk callback. + * + * The default behavior (AUTO) is to take the first matching entry in the clock + * table, which is intended to be the PLL based one if there is more than one. + * + * Setting the clock source using simple-card (clocks or + * system-clock-frequency property) sets clk_id = 0 = ADC3XXX_PLL_AUTO. + */ +#define ADC3XXX_PLL_AUTO 0 /* Use first available mode */ +#define ADC3XXX_PLL_ENABLE 1 /* Use PLL for clock generation */ +#define ADC3XXX_PLL_BYPASS 2 /* Don't use PLL for clock generation */ + +/* Register definitions. */ + +#define ADC3XXX_PAGE_SIZE 128 +#define ADC3XXX_REG(page, reg) ((page * ADC3XXX_PAGE_SIZE) + reg) + +/* + * Page 0 registers. + */ + +#define ADC3XXX_PAGE_SELECT ADC3XXX_REG(0, 0) +#define ADC3XXX_RESET ADC3XXX_REG(0, 1) + +/* 2-3 Reserved */ + +#define ADC3XXX_CLKGEN_MUX ADC3XXX_REG(0, 4) +#define ADC3XXX_PLL_PROG_PR ADC3XXX_REG(0, 5) +#define ADC3XXX_PLL_PROG_J ADC3XXX_REG(0, 6) +#define ADC3XXX_PLL_PROG_D_MSB ADC3XXX_REG(0, 7) +#define ADC3XXX_PLL_PROG_D_LSB ADC3XXX_REG(0, 8) + +/* 9-17 Reserved */ + +#define ADC3XXX_ADC_NADC ADC3XXX_REG(0, 18) +#define ADC3XXX_ADC_MADC ADC3XXX_REG(0, 19) +#define ADC3XXX_ADC_AOSR ADC3XXX_REG(0, 20) +#define ADC3XXX_ADC_IADC ADC3XXX_REG(0, 21) + +/* 23-24 Reserved */ + +#define ADC3XXX_CLKOUT_MUX ADC3XXX_REG(0, 25) +#define ADC3XXX_CLKOUT_M_DIV ADC3XXX_REG(0, 26) +#define ADC3XXX_INTERFACE_CTRL_1 ADC3XXX_REG(0, 27) +#define ADC3XXX_CH_OFFSET_1 ADC3XXX_REG(0, 28) +#define ADC3XXX_INTERFACE_CTRL_2 ADC3XXX_REG(0, 29) +#define ADC3XXX_BCLK_N_DIV ADC3XXX_REG(0, 30) +#define ADC3XXX_INTERFACE_CTRL_3 ADC3XXX_REG(0, 31) +#define ADC3XXX_INTERFACE_CTRL_4 ADC3XXX_REG(0, 32) +#define ADC3XXX_INTERFACE_CTRL_5 ADC3XXX_REG(0, 33) +#define ADC3XXX_I2S_SYNC ADC3XXX_REG(0, 34) +/* 35 Reserved */ +#define ADC3XXX_ADC_FLAG ADC3XXX_REG(0, 36) +#define ADC3XXX_CH_OFFSET_2 ADC3XXX_REG(0, 37) +#define ADC3XXX_I2S_TDM_CTRL ADC3XXX_REG(0, 38) +/* 39-41 Reserved */ +#define ADC3XXX_INTR_FLAG_1 ADC3XXX_REG(0, 42) +#define ADC3XXX_INTR_FLAG_2 ADC3XXX_REG(0, 43) +/* 44 Reserved */ +#define ADC3XXX_INTR_FLAG_ADC1 ADC3XXX_REG(0, 45) +/* 46 Reserved */ +#define ADC3XXX_INTR_FLAG_ADC2 ADC3XXX_REG(0, 47) +#define ADC3XXX_INT1_CTRL ADC3XXX_REG(0, 48) +#define ADC3XXX_INT2_CTRL ADC3XXX_REG(0, 49) +/* 50 Reserved */ +#define ADC3XXX_GPIO2_CTRL ADC3XXX_REG(0, 51) +#define ADC3XXX_GPIO1_CTRL ADC3XXX_REG(0, 52) +#define ADC3XXX_DOUT_CTRL ADC3XXX_REG(0, 53) +/* 54-56 Reserved */ +#define ADC3XXX_SYNC_CTRL_1 ADC3XXX_REG(0, 57) +#define ADC3XXX_SYNC_CTRL_2 ADC3XXX_REG(0, 58) +#define ADC3XXX_CIC_GAIN_CTRL ADC3XXX_REG(0, 59) +/* 60 Reserved */ +#define ADC3XXX_PRB_SELECT ADC3XXX_REG(0, 61) +#define ADC3XXX_INST_MODE_CTRL ADC3XXX_REG(0, 62) +/* 63-79 Reserved */ +#define ADC3XXX_MIC_POLARITY_CTRL ADC3XXX_REG(0, 80) +#define ADC3XXX_ADC_DIGITAL ADC3XXX_REG(0, 81) +#define ADC3XXX_ADC_FGA ADC3XXX_REG(0, 82) +#define ADC3XXX_LADC_VOL ADC3XXX_REG(0, 83) +#define ADC3XXX_RADC_VOL ADC3XXX_REG(0, 84) +#define ADC3XXX_ADC_PHASE_COMP ADC3XXX_REG(0, 85) +#define ADC3XXX_LEFT_CHN_AGC_1 ADC3XXX_REG(0, 86) +#define ADC3XXX_LEFT_CHN_AGC_2 ADC3XXX_REG(0, 87) +#define ADC3XXX_LEFT_CHN_AGC_3 ADC3XXX_REG(0, 88) +#define ADC3XXX_LEFT_CHN_AGC_4 ADC3XXX_REG(0, 89) +#define ADC3XXX_LEFT_CHN_AGC_5 ADC3XXX_REG(0, 90) +#define ADC3XXX_LEFT_CHN_AGC_6 ADC3XXX_REG(0, 91) +#define ADC3XXX_LEFT_CHN_AGC_7 ADC3XXX_REG(0, 92) +#define ADC3XXX_LEFT_AGC_GAIN ADC3XXX_REG(0, 93) +#define ADC3XXX_RIGHT_CHN_AGC_1 ADC3XXX_REG(0, 94) +#define ADC3XXX_RIGHT_CHN_AGC_2 ADC3XXX_REG(0, 95) +#define ADC3XXX_RIGHT_CHN_AGC_3 ADC3XXX_REG(0, 96) +#define ADC3XXX_RIGHT_CHN_AGC_4 ADC3XXX_REG(0, 97) +#define ADC3XXX_RIGHT_CHN_AGC_5 ADC3XXX_REG(0, 98) +#define ADC3XXX_RIGHT_CHN_AGC_6 ADC3XXX_REG(0, 99) +#define ADC3XXX_RIGHT_CHN_AGC_7 ADC3XXX_REG(0, 100) +#define ADC3XXX_RIGHT_AGC_GAIN ADC3XXX_REG(0, 101) +/* 102-127 Reserved */ + +/* + * Page 1 registers. + */ + +/* 1-25 Reserved */ +#define ADC3XXX_DITHER_CTRL ADC3XXX_REG(1, 26) +/* 27-50 Reserved */ +#define ADC3XXX_MICBIAS_CTRL ADC3XXX_REG(1, 51) +#define ADC3XXX_LEFT_PGA_SEL_1 ADC3XXX_REG(1, 52) +/* 53 Reserved */ +#define ADC3XXX_LEFT_PGA_SEL_2 ADC3XXX_REG(1, 54) +#define ADC3XXX_RIGHT_PGA_SEL_1 ADC3XXX_REG(1, 55) +#define ADC3XXX_RIGHT_PGA_SEL_2 ADC3XXX_REG(1, 57) +#define ADC3XXX_LEFT_APGA_CTRL ADC3XXX_REG(1, 59) +#define ADC3XXX_RIGHT_APGA_CTRL ADC3XXX_REG(1, 60) +#define ADC3XXX_LOW_CURRENT_MODES ADC3XXX_REG(1, 61) +#define ADC3XXX_ANALOG_PGA_FLAGS ADC3XXX_REG(1, 62) +/* 63-127 Reserved */ + +/* + * Register bits. + */ + +/* PLL Enable bits */ +#define ADC3XXX_ENABLE_PLL_SHIFT 7 +#define ADC3XXX_ENABLE_PLL (1 << ADC3XXX_ENABLE_PLL_SHIFT) +#define ADC3XXX_ENABLE_NADC_SHIFT 7 +#define ADC3XXX_ENABLE_NADC (1 << ADC3XXX_ENABLE_NADC_SHIFT) +#define ADC3XXX_ENABLE_MADC_SHIFT 7 +#define ADC3XXX_ENABLE_MADC (1 << ADC3XXX_ENABLE_MADC_SHIFT) +#define ADC3XXX_ENABLE_BCLK_SHIFT 7 +#define ADC3XXX_ENABLE_BCLK (1 << ADC3XXX_ENABLE_BCLK_SHIFT) + +/* Power bits */ +#define ADC3XXX_LADC_PWR_ON 0x80 +#define ADC3XXX_RADC_PWR_ON 0x40 + +#define ADC3XXX_SOFT_RESET 0x01 +#define ADC3XXX_BCLK_MASTER 0x08 +#define ADC3XXX_WCLK_MASTER 0x04 + +/* Interface register masks */ +#define ADC3XXX_FORMAT_MASK 0xc0 +#define ADC3XXX_FORMAT_SHIFT 6 +#define ADC3XXX_WLENGTH_MASK 0x30 +#define ADC3XXX_WLENGTH_SHIFT 4 +#define ADC3XXX_CLKDIR_MASK 0x0c +#define ADC3XXX_CLKDIR_SHIFT 2 + +/* Interface register bit patterns */ +#define ADC3XXX_FORMAT_I2S (0 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_DSP (1 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_RJF (2 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_LJF (3 << ADC3XXX_FORMAT_SHIFT) + +#define ADC3XXX_IFACE_16BITS (0 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_20BITS (1 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_24BITS (2 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_32BITS (3 << ADC3XXX_WLENGTH_SHIFT) + +/* PLL P/R bit offsets */ +#define ADC3XXX_PLLP_SHIFT 4 +#define ADC3XXX_PLLR_SHIFT 0 +#define ADC3XXX_PLL_PR_MASK 0x7f +#define ADC3XXX_PLLJ_MASK 0x3f +#define ADC3XXX_PLLD_MSB_MASK 0x3f +#define ADC3XXX_PLLD_LSB_MASK 0xff +#define ADC3XXX_NADC_MASK 0x7f +#define ADC3XXX_MADC_MASK 0x7f +#define ADC3XXX_AOSR_MASK 0xff +#define ADC3XXX_IADC_MASK 0xff +#define ADC3XXX_BDIV_MASK 0x7f + +/* PLL_CLKIN bits */ +#define ADC3XXX_PLL_CLKIN_SHIFT 2 +#define ADC3XXX_PLL_CLKIN_MCLK 0x0 +#define ADC3XXX_PLL_CLKIN_BCLK 0x1 +#define ADC3XXX_PLL_CLKIN_ZERO 0x3 + +/* CODEC_CLKIN bits */ +#define ADC3XXX_CODEC_CLKIN_SHIFT 0 +#define ADC3XXX_CODEC_CLKIN_MCLK 0x0 +#define ADC3XXX_CODEC_CLKIN_BCLK 0x1 +#define ADC3XXX_CODEC_CLKIN_PLL_CLK 0x3 + +#define ADC3XXX_USE_PLL ((ADC3XXX_PLL_CLKIN_MCLK << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_PLL_CLK << ADC3XXX_CODEC_CLKIN_SHIFT)) +#define ADC3XXX_NO_PLL ((ADC3XXX_PLL_CLKIN_ZERO << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_MCLK << ADC3XXX_CODEC_CLKIN_SHIFT)) + +/* Analog PGA control bits */ +#define ADC3XXX_LPGA_MUTE 0x80 +#define ADC3XXX_RPGA_MUTE 0x80 + +#define ADC3XXX_LPGA_GAIN_MASK 0x7f +#define ADC3XXX_RPGA_GAIN_MASK 0x7f + +/* ADC current modes */ +#define ADC3XXX_ADC_LOW_CURR_MODE 0x01 + +/* Left ADC Input selection bits */ +#define ADC3XXX_LCH_SEL1_SHIFT 0 +#define ADC3XXX_LCH_SEL2_SHIFT 2 +#define ADC3XXX_LCH_SEL3_SHIFT 4 +#define ADC3XXX_LCH_SEL4_SHIFT 6 + +#define ADC3XXX_LCH_SEL1X_SHIFT 0 +#define ADC3XXX_LCH_SEL2X_SHIFT 2 +#define ADC3XXX_LCH_SEL3X_SHIFT 4 +#define ADC3XXX_LCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_LPGA 0x80 + +/* Right ADC Input selection bits */ +#define ADC3XXX_RCH_SEL1_SHIFT 0 +#define ADC3XXX_RCH_SEL2_SHIFT 2 +#define ADC3XXX_RCH_SEL3_SHIFT 4 +#define ADC3XXX_RCH_SEL4_SHIFT 6 + +#define ADC3XXX_RCH_SEL1X_SHIFT 0 +#define ADC3XXX_RCH_SEL2X_SHIFT 2 +#define ADC3XXX_RCH_SEL3X_SHIFT 4 +#define ADC3XXX_RCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_RPGA 0x80 + +/* MICBIAS control bits */ +#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS1_SHIFT 5 +#define ADC3XXX_MICBIAS2_SHIFT 3 + +#define ADC3XXX_ADC_MAX_VOLUME 64 +#define ADC3XXX_ADC_POS_VOL 24 + +/* GPIO control bits (GPIO1_CTRL and GPIO2_CTRL) */ +#define ADC3XXX_GPIO_CTRL_CFG_MASK 0x3c +#define ADC3XXX_GPIO_CTRL_CFG_SHIFT 2 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK 0x01 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT 0 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_MASK 0x02 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_SHIFT 1 + +enum adc3xxx_type { + ADC3001 = 0, + ADC3101 +}; + +struct adc3xxx { + struct device *dev; + enum adc3xxx_type type; + struct clk *mclk; + struct regmap *regmap; + struct gpio_desc *rst_pin; + unsigned int pll_mode; + unsigned int sysclk; + unsigned int gpio_cfg[ADC3XXX_GPIOS_MAX]; /* value+1 (0 => not set) */ + unsigned int micbias_vg[ADC3XXX_MICBIAS_PINS]; + int master; + u8 page_no; + int use_pll; + struct gpio_chip gpio_chip; +}; + +static const unsigned int adc3xxx_gpio_ctrl_reg[ADC3XXX_GPIOS_MAX] = { + ADC3XXX_GPIO1_CTRL, + ADC3XXX_GPIO2_CTRL +}; + +static const unsigned int adc3xxx_micbias_shift[ADC3XXX_MICBIAS_PINS] = { + ADC3XXX_MICBIAS1_SHIFT, + ADC3XXX_MICBIAS2_SHIFT +}; + +static const struct reg_default adc3xxx_defaults[] = { + /* Page 0 */ + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x00 }, + { 4, 0x00 }, { 5, 0x11 }, { 6, 0x04 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x00 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x00 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x01 }, { 19, 0x01 }, + { 20, 0x80 }, { 21, 0x80 }, { 22, 0x04 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x01 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x02 }, { 30, 0x01 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x10 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x02 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x00 }, + { 44, 0x00 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x00 }, + { 52, 0x00 }, { 53, 0x12 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x00 }, { 59, 0x44 }, + { 60, 0x00 }, { 61, 0x01 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x00 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x00 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x88 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x7f }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x7f }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x00 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, { 110, 0x00 }, { 111, 0x00 }, + { 112, 0x00 }, { 113, 0x00 }, { 114, 0x00 }, { 115, 0x00 }, + { 116, 0x00 }, { 117, 0x00 }, { 118, 0x00 }, { 119, 0x00 }, + { 120, 0x00 }, { 121, 0x00 }, { 122, 0x00 }, { 123, 0x00 }, + { 124, 0x00 }, { 125, 0x00 }, { 126, 0x00 }, { 127, 0x00 }, + + /* Page 1 */ + { 128, 0x00 }, { 129, 0x00 }, { 130, 0x00 }, { 131, 0x00 }, + { 132, 0x00 }, { 133, 0x00 }, { 134, 0x00 }, { 135, 0x00 }, + { 136, 0x00 }, { 137, 0x00 }, { 138, 0x00 }, { 139, 0x00 }, + { 140, 0x00 }, { 141, 0x00 }, { 142, 0x00 }, { 143, 0x00 }, + { 144, 0x00 }, { 145, 0x00 }, { 146, 0x00 }, { 147, 0x00 }, + { 148, 0x00 }, { 149, 0x00 }, { 150, 0x00 }, { 151, 0x00 }, + { 152, 0x00 }, { 153, 0x00 }, { 154, 0x00 }, { 155, 0x00 }, + { 156, 0x00 }, { 157, 0x00 }, { 158, 0x00 }, { 159, 0x00 }, + { 160, 0x00 }, { 161, 0x00 }, { 162, 0x00 }, { 163, 0x00 }, + { 164, 0x00 }, { 165, 0x00 }, { 166, 0x00 }, { 167, 0x00 }, + { 168, 0x00 }, { 169, 0x00 }, { 170, 0x00 }, { 171, 0x00 }, + { 172, 0x00 }, { 173, 0x00 }, { 174, 0x00 }, { 175, 0x00 }, + { 176, 0x00 }, { 177, 0x00 }, { 178, 0x00 }, { 179, 0x00 }, + { 180, 0xff }, { 181, 0x00 }, { 182, 0x3f }, { 183, 0xff }, + { 184, 0x00 }, { 185, 0x3f }, { 186, 0x00 }, { 187, 0x80 }, + { 188, 0x80 }, { 189, 0x00 }, { 190, 0x00 }, { 191, 0x00 }, +}; + +static bool adc3xxx_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADC3XXX_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_range_cfg adc3xxx_ranges[] = { + { + .range_min = 0, + .range_max = 2 * ADC3XXX_PAGE_SIZE, + .selector_reg = ADC3XXX_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = ADC3XXX_PAGE_SIZE, + } +}; + +static const struct regmap_config adc3xxx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = adc3xxx_defaults, + .num_reg_defaults = ARRAY_SIZE(adc3xxx_defaults), + + .volatile_reg = adc3xxx_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + + .ranges = adc3xxx_ranges, + .num_ranges = ARRAY_SIZE(adc3xxx_ranges), + .max_register = 2 * ADC3XXX_PAGE_SIZE, +}; + +struct adc3xxx_rate_divs { + u32 mclk; + u32 rate; + u8 pll_p; + u8 pll_r; + u8 pll_j; + u16 pll_d; + u8 nadc; + u8 madc; + u8 aosr; +}; + +/* + * PLL and Clock settings. + * If p member is 0, PLL is not used. + * The order of the entries in this table have the PLL entries before + * the non-PLL entries, so that the PLL modes are preferred unless + * the PLL mode setting says otherwise. + */ +static const struct adc3xxx_rate_divs adc3xxx_divs[] = { + /* mclk, rate, p, r, j, d, nadc, madc, aosr */ + /* 8k rate */ + { 12000000, 8000, 1, 1, 7, 1680, 42, 2, 128 }, + { 12288000, 8000, 1, 1, 7, 0000, 42, 2, 128 }, + /* 11.025k rate */ + { 12000000, 11025, 1, 1, 6, 8208, 29, 2, 128 }, + /* 16k rate */ + { 12000000, 16000, 1, 1, 7, 1680, 21, 2, 128 }, + { 12288000, 16000, 1, 1, 7, 0000, 21, 2, 128 }, + /* 22.05k rate */ + { 12000000, 22050, 1, 1, 7, 560, 15, 2, 128 }, + /* 32k rate */ + { 12000000, 32000, 1, 1, 8, 1920, 12, 2, 128 }, + { 12288000, 32000, 1, 1, 8, 0000, 12, 2, 128 }, + /* 44.1k rate */ + { 12000000, 44100, 1, 1, 7, 5264, 8, 2, 128 }, + /* 48k rate */ + { 12000000, 48000, 1, 1, 7, 1680, 7, 2, 128 }, + { 12288000, 48000, 1, 1, 7, 0000, 7, 2, 128 }, + { 24576000, 48000, 1, 1, 3, 5000, 7, 2, 128 }, /* With PLL */ + { 24576000, 48000, 0, 0, 0, 0000, 2, 2, 128 }, /* Without PLL */ + /* 88.2k rate */ + { 12000000, 88200, 1, 1, 7, 5264, 4, 4, 64 }, + /* 96k rate */ + { 12000000, 96000, 1, 1, 8, 1920, 4, 4, 64 }, +}; + +static int adc3xxx_get_divs(struct device *dev, int mclk, int rate, int pll_mode) +{ + int i; + + dev_dbg(dev, "mclk = %d, rate = %d, clock mode %u\n", + mclk, rate, pll_mode); + for (i = 0; i < ARRAY_SIZE(adc3xxx_divs); i++) { + const struct adc3xxx_rate_divs *mode = &adc3xxx_divs[i]; + + /* Skip this entry if it doesn't fulfill the intended clock + * mode requirement. We consider anything besides the two + * modes below to be the same as ADC3XXX_PLL_AUTO. + */ + if ((pll_mode == ADC3XXX_PLL_BYPASS && mode->pll_p) || + (pll_mode == ADC3XXX_PLL_ENABLE && !mode->pll_p)) + continue; + + if (mode->rate == rate && mode->mclk == mclk) + return i; + } + + dev_info(dev, "Master clock rate %d and sample rate %d is not supported\n", + mclk, rate); + return -EINVAL; +} + +static int adc3xxx_pll_delay(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* 10msec delay needed after PLL power-up to allow + * PLL and dividers to stabilize (datasheet p13). + */ + usleep_range(10000, 20000); + + return 0; +} + +static const char * const adc_softstepping_text[] = { "1 step", "2 step", "off" }; +static SOC_ENUM_SINGLE_DECL(adc_softstepping_enum, ADC3XXX_ADC_DIGITAL, 0, + adc_softstepping_text); + +static const char * const multiplier_text[] = { "1", "2", "4", "8", "16", "32", "64", "128" }; +static SOC_ENUM_SINGLE_DECL(left_agc_attack_mult_enum, + ADC3XXX_LEFT_CHN_AGC_4, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(right_agc_attack_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_4, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(left_agc_decay_mult_enum, + ADC3XXX_LEFT_CHN_AGC_5, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(right_agc_decay_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_5, 0, multiplier_text); + +static const char * const dither_dc_offset_text[] = { + "0mV", "15mV", "30mV", "45mV", "60mV", "75mV", "90mV", "105mV", + "-15mV", "-30mV", "-45mV", "-60mV", "-75mV", "-90mV", "-105mV" +}; +static const unsigned int dither_dc_offset_values[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 9, 10, 11, 12, 13, 14, 15 +}; +static SOC_VALUE_ENUM_DOUBLE_DECL(dither_dc_offset_enum, + ADC3XXX_DITHER_CTRL, + 4, 0, 0xf, dither_dc_offset_text, + dither_dc_offset_values); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fine_tlv, -40, 10, 0); +/* AGC target: 8 values: -5.5, -8, -10, -12, -14, -17, -20, -24 dB */ +/* It would be nice to declare these in the order above, but empirically + * TLV_DB_SCALE_ITEM doesn't take lightly to the increment (second) parameter + * being negative, despite there being examples to the contrary in other + * drivers. So declare these in the order from lowest to highest, and + * set the invert flag in the SOC_DOUBLE_R_TLV declaration instead. + */ +static const DECLARE_TLV_DB_RANGE(agc_target_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-2400, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(-2000, 300, 0), + 4, 6, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 7, 7, TLV_DB_SCALE_ITEM(-550, 0, 0)); +/* Since the 'disabled' value (mute) is at the highest value in the dB + * range (i.e. just before -32 dB) rather than the lowest, we need to resort + * to using a TLV_DB_RANGE in order to get the mute value in the right place. + */ +static const DECLARE_TLV_DB_RANGE(agc_thresh_tlv, + 0, 30, TLV_DB_SCALE_ITEM(-9000, 200, 0), + 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +/* AGC hysteresis: 4 values: 1, 2, 4 dB, disabled (= mute) */ +static const DECLARE_TLV_DB_RANGE(agc_hysteresis_tlv, + 0, 1, TLV_DB_SCALE_ITEM(100, 100, 0), + 2, 2, TLV_DB_SCALE_ITEM(400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +static const DECLARE_TLV_DB_SCALE(agc_max_tlv, 0, 50, 0); +/* Input attenuation: -6 dB or 0 dB */ +static const DECLARE_TLV_DB_SCALE(input_attenuation_tlv, -600, 600, 0); + +static const struct snd_kcontrol_new adc3xxx_snd_controls[] = { + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADC3XXX_LEFT_APGA_CTRL, + ADC3XXX_RIGHT_APGA_CTRL, 0, 80, 0, pga_tlv), + SOC_DOUBLE("PGA Capture Switch", ADC3XXX_ADC_FGA, 7, 3, 1, 1), + SOC_DOUBLE_R("AGC Capture Switch", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_1, 7, 1, 0), + SOC_DOUBLE_R_TLV("AGC Target Level Capture Volume", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_2, 4, 0x07, 1, agc_target_tlv), + SOC_DOUBLE_R_TLV("AGC Noise Threshold Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 1, 0x1f, 1, agc_thresh_tlv), + SOC_DOUBLE_R_TLV("AGC Hysteresis Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 6, 3, 0, agc_hysteresis_tlv), + SOC_DOUBLE_R("AGC Clip Stepping Capture Switch", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 0, 1, 0), + /* + * Oddly enough, the data sheet says the default value + * for the left/right AGC maximum gain register field + * (ADC3XXX_LEFT/RIGHT_CHN_AGC_3 bits 0..6) is 0x7f = 127 + * (verified empirically) even though this value (indeed, above + * 0x50) is specified as 'Reserved. Do not use.' in the accompanying + * table in the data sheet. + */ + SOC_DOUBLE_R_TLV("AGC Maximum Capture Volume", ADC3XXX_LEFT_CHN_AGC_3, + ADC3XXX_RIGHT_CHN_AGC_3, 0, 0x50, 0, agc_max_tlv), + SOC_DOUBLE_R("AGC Attack Time", ADC3XXX_LEFT_CHN_AGC_4, + ADC3XXX_RIGHT_CHN_AGC_4, 3, 0x1f, 0), + /* Would like to have the multipliers as LR pairs, but there is + * no SOC_ENUM_foo which accepts two values in separate registers. + */ + SOC_ENUM("AGC Left Attack Time Multiplier", left_agc_attack_mult_enum), + SOC_ENUM("AGC Right Attack Time Multiplier", right_agc_attack_mult_enum), + SOC_DOUBLE_R("AGC Decay Time", ADC3XXX_LEFT_CHN_AGC_5, + ADC3XXX_RIGHT_CHN_AGC_5, 3, 0x1f, 0), + SOC_ENUM("AGC Left Decay Time Multiplier", left_agc_decay_mult_enum), + SOC_ENUM("AGC Right Decay Time Multiplier", right_agc_decay_mult_enum), + SOC_DOUBLE_R("AGC Noise Debounce", ADC3XXX_LEFT_CHN_AGC_6, + ADC3XXX_RIGHT_CHN_AGC_6, 0, 0x1f, 0), + SOC_DOUBLE_R("AGC Signal Debounce", ADC3XXX_LEFT_CHN_AGC_7, + ADC3XXX_RIGHT_CHN_AGC_7, 0, 0x0f, 0), + /* Read only register */ + SOC_DOUBLE_R_S_TLV("AGC Applied Capture Volume", ADC3XXX_LEFT_AGC_GAIN, + ADC3XXX_RIGHT_AGC_GAIN, 0, -24, 40, 6, 0, adc_tlv), + /* ADC soft stepping */ + SOC_ENUM("ADC Soft Stepping", adc_softstepping_enum), + /* Left/Right Input attenuation */ + SOC_SINGLE_TLV("Left Input IN_1L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_2L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2L_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_1L_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2R_3R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_2R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2R_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_1L_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2L_3L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_DOUBLE_R_S_TLV("ADC Volume Control Capture Volume", ADC3XXX_LADC_VOL, + ADC3XXX_RADC_VOL, 0, -24, 40, 6, 0, adc_tlv), + /* Empirically, the following doesn't work the way it's supposed + * to. Values 0, -0.1, -0.2 and -0.3 dB result in the same level, and + * -0.4 dB drops about 0.12 dB on a specific chip. + */ + SOC_DOUBLE_TLV("ADC Fine Volume Control Capture Volume", ADC3XXX_ADC_FGA, + 4, 0, 4, 1, adc_fine_tlv), + SOC_SINGLE("Left ADC Unselected CM Bias Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 6, 1, 0), + SOC_SINGLE("Right ADC Unselected CM Bias Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 6, 1, 0), + SOC_ENUM("Dither Control DC Offset", dither_dc_offset_enum), +}; + +/* Left input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Right input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Left Digital Mic input for left ADC */ +static const struct snd_kcontrol_new left_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Left ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 3, 0x1, 0), +}; + +/* Right Digital Mic input for Right ADC */ +static const struct snd_kcontrol_new right_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Right ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 2, 0x1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget adc3xxx_dapm_widgets[] = { + + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + /* PGA selection */ + SND_SOC_DAPM_PGA("Left PGA", ADC3XXX_LEFT_APGA_CTRL, 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", ADC3XXX_RIGHT_APGA_CTRL, 7, 1, NULL, 0), + + /* Digital Microphone Input Control for Left/Right ADC */ + SND_SOC_DAPM_MIXER("Left DMic Input", SND_SOC_NOPM, 0, 0, + &left_input_dmic_controls[0], + ARRAY_SIZE(left_input_dmic_controls)), + SND_SOC_DAPM_MIXER("Right DMic Input", SND_SOC_NOPM, 0, 0, + &right_input_dmic_controls[0], + ARRAY_SIZE(right_input_dmic_controls)), + + /* Left/Right ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ADC3XXX_ADC_DIGITAL, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ADC3XXX_ADC_DIGITAL, 6, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("IN_1L"), + SND_SOC_DAPM_INPUT("IN_1R"), + SND_SOC_DAPM_INPUT("IN_2L"), + SND_SOC_DAPM_INPUT("IN_2R"), + SND_SOC_DAPM_INPUT("IN_3L"), + SND_SOC_DAPM_INPUT("IN_3R"), + SND_SOC_DAPM_INPUT("DIFL_1L_1R"), + SND_SOC_DAPM_INPUT("DIFL_2L_3L"), + SND_SOC_DAPM_INPUT("DIFL_2R_3R"), + SND_SOC_DAPM_INPUT("DIFR_1L_1R"), + SND_SOC_DAPM_INPUT("DIFR_2L_3L"), + SND_SOC_DAPM_INPUT("DIFR_2R_3R"), + SND_SOC_DAPM_INPUT("DMic_L"), + SND_SOC_DAPM_INPUT("DMic_R"), + + /* Digital audio interface output */ + SND_SOC_DAPM_AIF_OUT("AIF_OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Clocks */ + SND_SOC_DAPM_SUPPLY("PLL_CLK", ADC3XXX_PLL_PROG_PR, ADC3XXX_ENABLE_PLL_SHIFT, + 0, adc3xxx_pll_delay, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("ADC_CLK", ADC3XXX_ADC_NADC, ADC3XXX_ENABLE_NADC_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_MOD_CLK", ADC3XXX_ADC_MADC, ADC3XXX_ENABLE_MADC_SHIFT, + 0, NULL, 0), + + /* This refers to the generated BCLK in master mode. */ + SND_SOC_DAPM_SUPPLY("BCLK", ADC3XXX_BCLK_N_DIV, ADC3XXX_ENABLE_BCLK_SHIFT, + 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route adc3xxx_intercon[] = { + /* Left input selection from switches */ + { "Left Input", "IN_1L Capture Switch", "IN_1L" }, + { "Left Input", "IN_2L Capture Switch", "IN_2L" }, + { "Left Input", "IN_3L Capture Switch", "IN_3L" }, + { "Left Input", "DIF_2L_3L Capture Switch", "DIFL_2L_3L" }, + { "Left Input", "DIF_1L_1R Capture Switch", "DIFL_1L_1R" }, + { "Left Input", "DIF_2R_3R Capture Switch", "DIFL_2R_3R" }, + { "Left Input", "IN_1R Capture Switch", "IN_1R" }, + + /* Left input selection to left PGA */ + { "Left PGA", NULL, "Left Input" }, + + /* Left PGA to left ADC */ + { "Left ADC", NULL, "Left PGA" }, + + /* Right input selection from switches */ + { "Right Input", "IN_1R Capture Switch", "IN_1R" }, + { "Right Input", "IN_2R Capture Switch", "IN_2R" }, + { "Right Input", "IN_3R Capture Switch", "IN_3R" }, + { "Right Input", "DIF_2R_3R Capture Switch", "DIFR_2R_3R" }, + { "Right Input", "DIF_1L_1R Capture Switch", "DIFR_1L_1R" }, + { "Right Input", "DIF_2L_3L Capture Switch", "DIFR_2L_3L" }, + { "Right Input", "IN_1L Capture Switch", "IN_1L" }, + + /* Right input selection to right PGA */ + { "Right PGA", NULL, "Right Input" }, + + /* Right PGA to right ADC */ + { "Right ADC", NULL, "Right PGA" }, + + /* Left DMic Input selection from switch */ + { "Left DMic Input", "Left ADC Capture Switch", "DMic_L" }, + + /* Left DMic to left ADC */ + { "Left ADC", NULL, "Left DMic Input" }, + + /* Right DMic Input selection from switch */ + { "Right DMic Input", "Right ADC Capture Switch", "DMic_R" }, + + /* Right DMic to right ADC */ + { "Right ADC", NULL, "Right DMic Input" }, + + /* ADC to AIF output */ + { "AIF_OUT", NULL, "Left ADC" }, + { "AIF_OUT", NULL, "Right ADC" }, + + /* Clocking */ + { "ADC_MOD_CLK", NULL, "ADC_CLK" }, + { "Left ADC", NULL, "ADC_MOD_CLK" }, + { "Right ADC", NULL, "ADC_MOD_CLK" }, + + { "BCLK", NULL, "ADC_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_pll_intercon[] = { + { "ADC_CLK", NULL, "PLL_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_bclk_out_intercon[] = { + { "AIF_OUT", NULL, "BCLK" } +}; + +static int adc3xxx_gpio_request(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + if (offset >= ADC3XXX_GPIOS_MAX) + return -EINVAL; + + /* GPIO1 is offset 0, GPIO2 is offset 1 */ + /* We check here that the GPIO pins are either not configured in the + * DT, or that they purposely are set as outputs. + * (Input mode not yet implemented). + */ + if (adc3xxx->gpio_cfg[offset] != 0 && + adc3xxx->gpio_cfg[offset] != ADC3XXX_GPIO_GPO + 1) + return -EINVAL; + + return 0; +} + +static int adc3xxx_gpio_direction_out(struct gpio_chip *chip, + unsigned int offset, int value) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + /* Set GPIO output function. */ + return regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[offset], + ADC3XXX_GPIO_CTRL_CFG_MASK | + ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK, + ADC3XXX_GPIO_GPO << ADC3XXX_GPIO_CTRL_CFG_SHIFT | + !!value << ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT); +} + +/* With only GPIO outputs configured, we never get the .direction_out call, + * so we set the output mode and output value in the same call. Hence + * .set in practice does the same thing as .direction_out . + */ +static void adc3xxx_gpio_set(struct gpio_chip *chip, unsigned int offset, + int value) +{ + (void) adc3xxx_gpio_direction_out(chip, offset, value); +} + +/* Even though we only support GPIO output for now, some GPIO clients + * want to read the current pin state using the .get callback. + */ +static int adc3xxx_gpio_get(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + unsigned int regval; + int ret; + + /* We only allow output pins, so just read the value set in the output + * pin register field. + */ + ret = regmap_read(adc3xxx->regmap, adc3xxx_gpio_ctrl_reg[offset], &regval); + if (ret) + return ret; + return !!(regval & ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK); +} + +static const struct gpio_chip adc3xxx_gpio_chip = { + .label = "adc3xxx", + .owner = THIS_MODULE, + .request = adc3xxx_gpio_request, + .direction_output = adc3xxx_gpio_direction_out, + .set = adc3xxx_gpio_set, + .get = adc3xxx_gpio_get, + .can_sleep = 1, +}; + +static void adc3xxx_free_gpio(struct adc3xxx *adc3xxx) +{ + gpiochip_remove(&adc3xxx->gpio_chip); +} + +static void adc3xxx_init_gpio(struct adc3xxx *adc3xxx) +{ + int gpio, micbias; + int ret; + + adc3xxx->gpio_chip = adc3xxx_gpio_chip; + adc3xxx->gpio_chip.ngpio = ADC3XXX_GPIOS_MAX; + adc3xxx->gpio_chip.parent = adc3xxx->dev; + adc3xxx->gpio_chip.base = -1; + + ret = gpiochip_add_data(&adc3xxx->gpio_chip, adc3xxx); + if (ret) + dev_err(adc3xxx->dev, "Failed to add gpios: %d\n", ret); + + /* Set up potential GPIO configuration from the devicetree. + * This allows us to set up things which are not software + * controllable GPIOs, such as PDM microphone I/O, + */ + for (gpio = 0; gpio < ADC3XXX_GPIOS_MAX; gpio++) { + unsigned int cfg = adc3xxx->gpio_cfg[gpio]; + + if (cfg) { + cfg--; /* actual value to use is stored +1 */ + regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[gpio], + ADC3XXX_GPIO_CTRL_CFG_MASK, + cfg << ADC3XXX_GPIO_CTRL_CFG_SHIFT); + } + } + + /* Set up micbias voltage */ + for (micbias = 0; micbias < ADC3XXX_MICBIAS_PINS; micbias++) { + unsigned int vg = adc3xxx->micbias_vg[micbias]; + + regmap_update_bits(adc3xxx->regmap, + ADC3XXX_MICBIAS_CTRL, + ADC3XXX_MICBIAS_MASK << adc3xxx_micbias_shift[micbias], + vg << adc3xxx_micbias_shift[micbias]); + } +} + +static int adc3xxx_parse_dt_gpio(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *cfg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val & ~15 || val == 7 || val >= 11) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + if (val == ADC3XXX_GPIO_GPI) + dev_warn(dev, "GPIO Input read not yet implemented\n"); + *cfg = val + 1; /* 0 => not set up, all others shifted +1 */ + } + return 0; +} + +static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *vg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val >= ADC3XXX_MICBIAS_AVDD) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + *vg = val; + } + return 0; +} + +static int adc3xxx_parse_pll_mode(uint32_t val, unsigned int *pll_mode) +{ + if (val != ADC3XXX_PLL_ENABLE && val != ADC3XXX_PLL_BYPASS && + val != ADC3XXX_PLL_AUTO) + return -EINVAL; + + *pll_mode = val; + + return 0; +} + +static void adc3xxx_setup_pll(struct snd_soc_component *component, + int div_entry) +{ + int i = div_entry; + + /* P & R values */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, + (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | + (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); + /* J value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, + adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); + /* D value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, + adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, + (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); +} + +static int adc3xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(dai->component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int i, width = 16; + u8 iface_len, bdiv; + + i = adc3xxx_get_divs(component->dev, adc3xxx->sysclk, + params_rate(params), adc3xxx->pll_mode); + + if (i < 0) + return i; + + /* select data word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface_len = ADC3XXX_IFACE_16BITS; + width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_len = ADC3XXX_IFACE_20BITS; + width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_len = ADC3XXX_IFACE_24BITS; + width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_len = ADC3XXX_IFACE_32BITS; + width = 32; + break; + default: + dev_err(component->dev, "Unsupported serial data format\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_WLENGTH_MASK, iface_len); + if (adc3xxx_divs[i].pll_p) { /* If PLL used for this mode */ + adc3xxx_setup_pll(component, i); + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_USE_PLL); + if (!adc3xxx->use_pll) { + snd_soc_dapm_add_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 1; + } + } else { + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_NO_PLL); + if (adc3xxx->use_pll) { + snd_soc_dapm_del_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 0; + } + } + + /* NADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_NADC, + ADC3XXX_NADC_MASK, adc3xxx_divs[i].nadc); + /* MADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_MADC, + ADC3XXX_MADC_MASK, adc3xxx_divs[i].madc); + /* AOSR */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_AOSR, + ADC3XXX_AOSR_MASK, adc3xxx_divs[i].aosr); + /* BDIV N Value */ + /* BCLK is (by default) set up to be derived from ADC_CLK */ + bdiv = (adc3xxx_divs[i].aosr * adc3xxx_divs[i].madc) / (2 * width); + snd_soc_component_update_bits(component, ADC3XXX_BCLK_N_DIV, + ADC3XXX_BDIV_MASK, bdiv); + + return 0; +} + +static const char *adc3xxx_pll_mode_text(int pll_mode) +{ + switch (pll_mode) { + case ADC3XXX_PLL_AUTO: + return "PLL auto"; + case ADC3XXX_PLL_ENABLE: + return "PLL enable"; + case ADC3XXX_PLL_BYPASS: + return "PLL bypass"; + default: + break; + } + + return "PLL unknown"; +} + +static int adc3xxx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int ret; + + ret = adc3xxx_parse_pll_mode(clk_id, &adc3xxx->pll_mode); + if (ret < 0) + return ret; + + adc3xxx->sysclk = freq; + dev_dbg(component->dev, "Set sysclk to %u Hz, %s\n", + freq, adc3xxx_pll_mode_text(adc3xxx->pll_mode)); + return 0; +} + +static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + u8 clkdir = 0, format = 0; + int master = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + master = 1; + clkdir = ADC3XXX_BCLK_MASTER | ADC3XXX_WCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBC_CFC: + master = 0; + break; + default: + dev_err(component->dev, "Invalid DAI clock setup\n"); + return -EINVAL; + } + + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_RJF; + break; + case SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_LJF; + break; + default: + dev_err(component->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* Add/del route enabling BCLK output as applicable */ + if (master && !adc3xxx->master) + snd_soc_dapm_add_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + else if (!master && adc3xxx->master) + snd_soc_dapm_del_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + adc3xxx->master = master; + + /* set clock direction and format */ + return snd_soc_component_update_bits(component, + ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_CLKDIR_MASK | ADC3XXX_FORMAT_MASK, + clkdir | format); +} + +static const struct snd_soc_dai_ops adc3xxx_dai_ops = { + .hw_params = adc3xxx_hw_params, + .set_sysclk = adc3xxx_set_dai_sysclk, + .set_fmt = adc3xxx_set_dai_fmt, +}; + +static struct snd_soc_dai_driver adc3xxx_dai = { + .name = "tlv320adc3xxx-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ADC3XXX_RATES, + .formats = ADC3XXX_FORMATS, + }, + .ops = &adc3xxx_dai_ops, +}; + +static const struct snd_soc_component_driver soc_component_dev_adc3xxx = { + .controls = adc3xxx_snd_controls, + .num_controls = ARRAY_SIZE(adc3xxx_snd_controls), + .dapm_widgets = adc3xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adc3xxx_dapm_widgets), + .dapm_routes = adc3xxx_intercon, + .num_dapm_routes = ARRAY_SIZE(adc3xxx_intercon), +}; + +static int adc3xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct adc3xxx *adc3xxx = NULL; + int ret; + + adc3xxx = devm_kzalloc(dev, sizeof(struct adc3xxx), GFP_KERNEL); + if (!adc3xxx) + return -ENOMEM; + adc3xxx->dev = dev; + + adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(adc3xxx->rst_pin)) { + return dev_err_probe(dev, PTR_ERR(adc3xxx->rst_pin), + "Failed to request rst_pin\n"); + } + + adc3xxx->mclk = devm_clk_get(dev, NULL); + if (IS_ERR(adc3xxx->mclk)) { + /* + * The chip itself supports running off the BCLK either + * directly or via the PLL, but the driver does not (yet), so + * having a specified mclk is required. Otherwise, we could + * use the lack of a clocks property to indicate when BCLK is + * intended as the clock source. + */ + return dev_err_probe(dev, PTR_ERR(adc3xxx->mclk), + "Failed to acquire MCLK\n"); + } else if (adc3xxx->mclk) { + ret = clk_prepare_enable(adc3xxx->mclk); + if (ret < 0) + return ret; + dev_dbg(dev, "Enabled MCLK, freq %lu Hz\n", clk_get_rate(adc3xxx->mclk)); + } + + ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio_cfg[0]); + if (ret < 0) + return ret; + ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio_cfg[1]); + if (ret < 0) + return ret; + ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias1-vg", &adc3xxx->micbias_vg[0]); + if (ret < 0) + return ret; + ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias2-vg", &adc3xxx->micbias_vg[1]); + if (ret < 0) + return ret; + + adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); + if (IS_ERR(adc3xxx->regmap)) { + ret = PTR_ERR(adc3xxx->regmap); + return ret; + } + + i2c_set_clientdata(i2c, adc3xxx); + + adc3xxx->type = id->driver_data; + + /* Reset codec chip */ + gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); + usleep_range(2000, 100000); /* Requirement: > 10 ns (datasheet p13) */ + gpiod_set_value_cansleep(adc3xxx->rst_pin, 0); + + /* Potentially set up pins used as GPIOs */ + adc3xxx_init_gpio(adc3xxx); + + ret = snd_soc_register_component(dev, + &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int __exit adc3xxx_i2c_remove(struct i2c_client *client) +{ + struct adc3xxx *adc3xxx = i2c_get_clientdata(client); + + if (adc3xxx->mclk) + clk_disable_unprepare(adc3xxx->mclk); + adc3xxx_free_gpio(adc3xxx); + snd_soc_unregister_component(&client->dev); + return 0; +} + +static const struct of_device_id tlv320adc3xxx_of_match[] = { + { .compatible = "ti,tlv320adc3001", }, + { .compatible = "ti,tlv320adc3101", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adc3xxx_of_match); + +static const struct i2c_device_id adc3xxx_i2c_id[] = { + { "tlv320adc3001", ADC3001 }, + { "tlv320adc3101", ADC3101 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, adc3xxx_i2c_id); + +static struct i2c_driver adc3xxx_i2c_driver = { + .driver = { + .name = "tlv320adc3xxx-codec", + .of_match_table = tlv320adc3xxx_of_match, + }, + .probe = adc3xxx_i2c_probe, + .remove = adc3xxx_i2c_remove, + .id_table = adc3xxx_i2c_id, +}; + +module_i2c_driver(adc3xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320ADC3xxx codec driver"); +MODULE_AUTHOR("shahina.s(a)mistralsolutions.com"); +MODULE_LICENSE("GPL v2"); -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[PATCH 0/5] ASoC: Intel: catpt: Dma-transfer fix and couple
by Cezary Rojewski 21 Dec '21

21 Dec '21

Set is made of one fix for dma-transfer so that result of dmaengine_submit() is tested before moving on, and few cleanups: - two non-impactful, where catpt_component_open() layout gets improved slightly as well as relocation of couple of locals found in PCM-functions so that they look more cohesive - no need to expose catpt-driver board-matching information globally. Most fields are not by it and it's the sole user of haswell_machines table. By having them locally it is clear what is actually being used Cezary Rojewski (5): ASoC: Intel: catpt: Test dmaengine_submit() result before moving on ASoC: Intel: catpt: Reduce size of catpt_component_open() ASoC: Intel: catpt: Streamline locals declaration for PCM-functions ASoC: Intel: catpt: Drop SND_SOC_ACPI_INTEL_MATCH dependency ASoC: Intel: Drop legacy HSW/BDW board-match information include/sound/soc-acpi-intel-match.h | 1 - sound/soc/intel/Kconfig | 2 +- sound/soc/intel/catpt/device.c | 33 +++++++++++++++-- sound/soc/intel/catpt/dsp.c | 14 ++++++- sound/soc/intel/catpt/pcm.c | 37 +++++++++---------- .../common/soc-acpi-intel-hsw-bdw-match.c | 16 -------- 6 files changed, 61 insertions(+), 42 deletions(-) -- 2.25.1

3 14

[PATCH] ASoC: sh: rz-ssi: Check return value of pm_runtime_resume_and_get()
by Heiner Kallweit 21 Dec '21

21 Dec '21

The return value of pm_runtime_resume_and_get() needs to be checked to avoid a usage count imbalance in the error case. This fix is basically the same as 92c959bae2e5 ("reset: renesas: Fix Runtime PM usage"), and the last step before pm_runtime_resume_and_get() can be annotated as __must_check. Fixes: 03e786bd4341 ("ASoC: sh: Add RZ/G2L SSIF-2 driver") Cc: stable(a)vger.kernel.org Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> --- sound/soc/sh/rz-ssi.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index fa0cc08f7..850387dd8 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1020,7 +1020,12 @@ static int rz_ssi_probe(struct platform_device *pdev) reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); - pm_runtime_resume_and_get(&pdev->dev); + ret = pm_runtime_resume_and_get(&pdev->dev); + if (ret < 0) { + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); + return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + } spin_lock_init(&ssi->lock); dev_set_drvdata(&pdev->dev, ssi); -- 2.34.1

2 1

[PATCH v4 00/22] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 21 Dec '21

21 Dec '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. I based S/PDIF patches on Arnd's Bergmann patch from a separate series [1] that removes obsolete slave_id. This eases merging of the patches by removing the merge conflict. This is a note for Mark Brown. I also based this series on top of power management series [2]. I.e. [2] should be applied first, otherwise "Add S/PDIF node to Tegra20 device-tree" patch should have merge conflict. This is a note for Thierry. [1] https://patchwork.ozlabs.org/project/linux-tegra/list/?series=273312 [2] https://patchwork.ozlabs.org/project/linux-tegra/list/?series=274534 Changelog: v4: - Added patches that update multi_v7_defconfig with the enabled S/PDIF and APB DMA drivers. v3: - Renamed S/PDIF device-tree clocks as was suggested by Rob Herring. - Added r-bs and acks that were given by Rob Herring to v2. v2: - Corrected I2S yaml problem that was reported by the DT bot for v1 by removing the non-existent required clock-names property. - Removed assigned-clocks property from S/PDIF yaml since this property is now inherited from the clocks property. - Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making it the first sound/soc patch in the series, like it was suggested by Mark Brown in the comment to v1. Also reworded commit message of this patch to *not* make it looks like it should be backported to stable kernels. Arnd Bergmann (1): ASoC: tegra20-spdif: stop setting slave_id Dmitry Osipenko (21): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property dt-bindings: host1x: Document optional HDMI sound-dai-cells ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates drm/tegra: hdmi: Unwind tegra_hdmi_init() errors drm/tegra: hdmi: Register audio CODEC on Tegra20 ARM: tegra_defconfig: Enable S/PDIF driver ARM: config: multi v7: Enable NVIDIA Tegra20 S/PDIF driver ARM: config: multi v7: Enable NVIDIA Tegra20 APB DMA driver ARM: tegra: Add S/PDIF node to Tegra20 device-tree ARM: tegra: Add HDMI audio graph to Tegra20 device-tree ARM: tegra: acer-a500: Enable S/PDIF and HDMI audio ARM: tegra: paz00: Enable S/PDIF and HDMI audio .../display/tegra/nvidia,tegra20-host1x.txt | 1 + .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 77 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 85 ++++++++ .../boot/dts/tegra20-acer-a500-picasso.dts | 8 + arch/arm/boot/dts/tegra20-paz00.dts | 8 + arch/arm/boot/dts/tegra20.dtsi | 40 +++- arch/arm/configs/multi_v7_defconfig | 2 + arch/arm/configs/tegra_defconfig | 1 + drivers/gpu/drm/tegra/Kconfig | 3 + drivers/gpu/drm/tegra/hdmi.c | 168 +++++++++++++-- sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 197 ++++++++++++------ sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 16 files changed, 574 insertions(+), 103 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

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[PATCH 0/3] ASoC: SOF: couple of cleanups
by Pierre-Louis Bossart 21 Dec '21

21 Dec '21

Guennadi spotted inconsistencies with our 'const' handling, Ajit Kumar flagged a missing check for a null pointer and we missed the definition of debug zones. This should be queued for 5.17. Ajit Kumar Pandey (1): ASoC: SOF: ipc: Add null pointer check for substream->runtime Guennadi Liakhovetski (1): ASoC: SOF: avoid casting "const" attribute away Karol Trzcinski (1): ipc: debug: Add shared memory heap to memory scan include/sound/sof/debug.h | 2 ++ sound/soc/sof/amd/renoir.c | 7 +++-- sound/soc/sof/intel/atom.c | 13 +++++---- sound/soc/sof/intel/atom.h | 4 +-- sound/soc/sof/intel/bdw.c | 11 +++---- sound/soc/sof/intel/hda.c | 60 ++++++++++++++++++-------------------- sound/soc/sof/intel/hda.h | 4 +-- sound/soc/sof/ipc.c | 3 +- sound/soc/sof/ops.h | 8 +++-- sound/soc/sof/sof-audio.c | 9 +++--- sound/soc/sof/sof-priv.h | 4 +-- 11 files changed, 66 insertions(+), 59 deletions(-) -- 2.25.1

2 4

[PATCH 0/3] ASoC: SOF: remove suport for TRIGGER_RESUME
by Pierre-Louis Bossart 21 Dec '21

21 Dec '21

None of the SOF platforms support INFO_RESUME, and rely on the indirect path used by the ALSA core with the prepare and TRIGGER_START steps. Let's remove the left-over dead code. Ranjani Sridharan (3): ASoC: SOF: pcm: remove support for RESUME trigger ASoC: SOF: Intel: hda: remove support for RESUME trigger ASoC: SOF: Intel: hda: remove support for RESUME in platform trigger sound/soc/sof/intel/hda-dai.c | 10 ---------- sound/soc/sof/intel/hda-stream.c | 1 - sound/soc/sof/pcm.c | 20 -------------------- 3 files changed, 31 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: SOF: Kconfig: Make the SOF_DEVELOPER_SUPPORT depend on SND_SOC_SOF
by Pierre-Louis Bossart 21 Dec '21

21 Dec '21

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> SND_SOC_SOF_DEVELOPER_SUPPORT contains options affecting how the built SOF driver stack will behave, enables debug options and other features. These options have no meaning if the SND_SOC_SOF is not even enabled. If we have SOF client options under developer_support and debug they can be selected to be built even without the core, but they do need symbols from the core (the sof-client API) which can result build failure. In Kconfig we can have SND_SOC_SOF_TOPLEVEL=y SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST=y for example, which will make the flood client to be built, but the SOF core is not as SND_SOC_SOF is not selected. Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.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 ac34c330cf0c..1a7d6cefd3b7 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SOF_DEBUG_PROBES config SND_SOC_SOF_DEVELOPER_SUPPORT bool "SOF developer options support" - depends on EXPERT + depends on EXPERT && SND_SOC_SOF help This option unlocks SOF developer options for debug/performance/ code hardening. -- 2.25.1

2 1

[PATCH v2] ASoC: dt-bindings: audio-graph-port: enable both flag/phandle for bitclock/frame-master
by Kuninori Morimoto 21 Dec '21

21 Dec '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_daifmt_parse_clock_provider_raw() is handling both bitclock/frame-master, and is supporting both flag/phandle. Current DT is assuming it is flag style. This patch allows both case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - anyOf -> oneOf .../devicetree/bindings/sound/audio-graph-port.yaml | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 43e7f86e3b23..476dcb49ece6 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -42,10 +42,15 @@ patternProperties: $ref: /schemas/types.yaml#/definitions/flag frame-master: description: Indicates dai-link frame master. - $ref: /schemas/types.yaml#/definitions/phandle + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle bitclock-master: description: Indicates dai-link bit clock master - $ref: /schemas/types.yaml#/definitions/phandle + oneOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + dai-format: description: audio format. items: -- 2.25.1

3 2

[PATCH 00/22] ASoC: Use dev_err_probe() helper
by Kuninori Morimoto 21 Dec '21

21 Dec '21

I'm now posting below patch to ASoC/DT ML, sorry for my multi post, but this patch-set is independent from it. ASoC: dt-bindings: audio-graph-port: enable both flag/phandle for bitclock/frame-master https://lore.kernel.org/r/20211208064852.507977-1-kuninori.morimoto.gx@rene… This patch-set tries to use dev_err_probe() helper function instead of manual dev_err() code. I hope ASoC will be more clean code by this patch-set. - if (err < 0) { - if (err != -EPROBE_DEFER) - dev_err(...); - return err; - } + if (err < 0) + return dev_err_probe(...); Kuninori Morimoto (22): ASoC: codecs: ak4118: Use dev_err_probe() helper ASoC: codecs: es7241: Use dev_err_probe() helper ASoC: codecs: max9759: Use dev_err_probe() helper ASoC: codecs: max9860: Use dev_err_probe() helper ASoC: codecs: pcm3168a: Use dev_err_probe() helper ASoC: codecs: sgtl5000: Use dev_err_probe() helper ASoC: codecs: simple-amplifier: Use dev_err_probe() helper ASoC: codecs: simple-mux: Use dev_err_probe() helper ASoC: codecs: ssm2305: Use dev_err_probe() helper ASoC: codecs: tlv320aic31xx: Use dev_err_probe() helper ASoC: ateml: Use dev_err_probe() helper ASoC: ti: Use dev_err_probe() helper ASoC: fsl: Use dev_err_probe() helper ASoC: generic: Use dev_err_probe() helper ASoC: img: Use dev_err_probe() helper ASoC: meson: Use dev_err_probe() helper ASoC: mxs: Use dev_err_probe() helper ASoC: qcom: Use dev_err_probe() helper ASoC: rockchip: Use dev_err_probe() helper ASoC: samsung: Use dev_err_probe() helper ASoC: stm: Use dev_err_probe() helper ASoC: sunxi: Use dev_err_probe() helper sound/soc/atmel/mikroe-proto.c | 6 +-- sound/soc/atmel/tse850-pcm5142.c | 32 +++++-------- sound/soc/codecs/ak4118.c | 18 +++---- sound/soc/codecs/es7241.c | 28 ++++------- sound/soc/codecs/max9759.c | 28 ++++------- sound/soc/codecs/max9860.c | 12 ++--- sound/soc/codecs/pcm3168a.c | 22 +++------ sound/soc/codecs/sgtl5000.c | 5 +- sound/soc/codecs/simple-amplifier.c | 10 ++-- sound/soc/codecs/simple-mux.c | 10 ++-- sound/soc/codecs/ssm2305.c | 11 ++--- sound/soc/codecs/tlv320aic31xx.c | 16 ++----- sound/soc/fsl/fsl-asoc-card.c | 3 +- sound/soc/fsl/imx-card.c | 17 +++---- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-spdif.c | 4 +- sound/soc/generic/audio-graph-card.c | 5 +- sound/soc/generic/audio-graph-card2.c | 4 +- sound/soc/generic/simple-card.c | 3 +- sound/soc/img/img-i2s-in.c | 8 ++-- sound/soc/img/img-i2s-out.c | 24 ++++------ sound/soc/img/img-parallel-out.c | 24 ++++------ sound/soc/img/img-spdif-in.c | 8 ++-- sound/soc/img/img-spdif-out.c | 24 ++++------ sound/soc/img/pistachio-internal-dac.c | 9 ++-- sound/soc/meson/aiu.c | 36 +++++--------- sound/soc/meson/axg-fifo.c | 16 ++----- sound/soc/meson/axg-pdm.c | 25 +++------- sound/soc/meson/axg-spdifin.c | 17 ++----- sound/soc/meson/axg-spdifout.c | 17 ++----- sound/soc/meson/axg-tdm-formatter.c | 50 ++++++-------------- sound/soc/meson/axg-tdm-interface.c | 25 +++------- sound/soc/meson/meson-card-utils.c | 8 ++-- sound/soc/meson/t9015.c | 14 ++---- sound/soc/mxs/mxs-sgtl5000.c | 8 +--- sound/soc/qcom/common.c | 10 ++-- sound/soc/rockchip/rk3288_hdmi_analog.c | 10 ++-- sound/soc/samsung/aries_wm8994.c | 17 +++---- sound/soc/samsung/arndale.c | 5 +- sound/soc/samsung/littlemill.c | 5 +- sound/soc/samsung/lowland.c | 5 +- sound/soc/samsung/odroid.c | 4 +- sound/soc/samsung/smdk_wm8994.c | 4 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 +- sound/soc/samsung/snow.c | 9 ++-- sound/soc/samsung/speyside.c | 5 +- sound/soc/samsung/tm2_wm5110.c | 3 +- sound/soc/samsung/tobermory.c | 5 +- sound/soc/stm/stm32_i2s.c | 62 +++++++++---------------- sound/soc/stm/stm32_sai.c | 37 ++++++--------- sound/soc/stm/stm32_sai_sub.c | 25 ++++------ sound/soc/stm/stm32_spdifrx.c | 44 ++++++------------ sound/soc/sunxi/sun4i-codec.c | 3 +- sound/soc/ti/davinci-mcasp.c | 10 ++-- sound/soc/ti/j721e-evm.c | 10 ++-- 55 files changed, 271 insertions(+), 557 deletions(-) -- 2.25.1

2 23

[PATCH] ASoC: mediatek: mt8195: correct default value
by Trevor Wu 21 Dec '21

21 Dec '21

mt8195_cg_patch is used to reset the default value of audio cg, so the register value could be consistent with CCF reference count. Nevertheless, AUDIO_TOP_CON1[1:0] is used to control an internal mux, and it's expected to keep the default value 0. This patch corrects the default value in case an unexpected behavior happens in the future. Fixes: 6746cc8582599 ("ASoC: mediatek: mt8195: add platform driver") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 8a6db24116e3..e425f868476a 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3029,7 +3029,7 @@ static const struct reg_sequence mt8195_afe_reg_defaults[] = { static const struct reg_sequence mt8195_cg_patch[] = { { AUDIO_TOP_CON0, 0xfffffffb }, - { AUDIO_TOP_CON1, 0xfffffffa }, + { AUDIO_TOP_CON1, 0xfffffff8 }, }; static int mt8195_afe_init_registers(struct mtk_base_afe *afe) -- 2.18.0

2 1

RE: [PATCH] sound: fsl: add missing put_device() call in imx_hdmi_probe()
by 王擎 20 Dec '21

20 Dec '21

>> From: Wang Qing <wangqing(a)vivo.com> >> >> of_find_device_by_node() takes a reference to the embedded struct device >> which needs to be dropped when error return. > >... > >> data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); >> if (!data) { >> + put_device(&cpu_pdev->dev); > >If it's of_find_device_by_node() you need an of_node_put() since you're >dropping a reference on the OF node. > Label fail will drop a reference on the OF node. Also, put_device() is called later except this branch, we just need to add put_device() here. Thanks, Qing >> ret = -ENOMEM; >> goto fail;

2 1

[PATCH 0/3] Fix Tegra194 HDA regression
by Sameer Pujar 20 Dec '21

20 Dec '21

HDA probe failure is observed on Tegra194 based platforms and this happens due to a reset failure. The series fixes this problem by avoiding explicit resets on BPMP based devices. Sameer Pujar (3): ALSA: hda/tegra: Skip reset on BPMP devices dt-bindings: sound: tegra: Update HDA resets arm64: tegra: Remove non existent Tegra194 reset .../bindings/sound/nvidia,tegra30-hda.yaml | 13 ++++++++---- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 5 ++--- sound/pci/hda/hda_tegra.c | 24 +++++++++++++++++----- 3 files changed, 30 insertions(+), 12 deletions(-) -- 2.7.4

5 39

Missing profile 7.1
by GitHub issues - opened 19 Dec '21

19 Dec '21

alsa-project/alsa-utils issue #128 was opened from raven2cz: Hello, I have several years profile 7.1 which is missing now around 1.5-2months. I already write these issues * https://bbs.archlinux.org/viewtopic.php?id=272280 * https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1309 It seems that the problem can be in alsa-utils. ``` control.45 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 comment { access read type INTEGER count 6 range '0 - 36' } } ``` I already try to change kernels, not help. See inside the issues whole test env. Can you help me? Issue URL : https://github.com/alsa-project/alsa-utils/issues/128 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 1/3] ASoC: dt-bindings: Use name-prefix schema
by Alexander Stein 18 Dec '21

18 Dec '21

name-prefix.txt does not exist anymore, just reference the schema instead. Signed-off-by: Alexander Stein <alexander.stein(a)mailbox.org> --- .../devicetree/bindings/sound/simple-audio-amplifier.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml index 26379377a7ac..1f0327474744 100644 --- a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -9,6 +9,9 @@ title: Simple Audio Amplifier Device Tree Bindings maintainers: - Jerome Brunet <jbrunet(a)baylibre.com> +allOf: + - $ref: name-prefix.yaml# + properties: compatible: enum: @@ -22,10 +25,7 @@ properties: description: > power supply for the device - sound-name-prefix: - $ref: /schemas/types.yaml#/definitions/string - description: > - See ./name-prefix.txt + sound-name-prefix: pass required: - compatible -- 2.34.1

1 2

Re: [syzbot] KMSAN: uninit-value in snd_rawmidi_ioctl
by Jaroslav Kysela 18 Dec '21

18 Dec '21

On 18. 12. 21 12:17, syzbot wrote: > Hello, > > syzbot found the following issue on: > > HEAD commit: 8b936c96768e kmsan: core: remove the accidentally committe.. > git tree: https://github.com/google/kmsan.git master > console output: https://syzkaller.appspot.com/x/log.txt?x=11791d89b00000 > kernel config: https://syzkaller.appspot.com/x/.config?x=e00a8959fdd3f3e8 > dashboard link: https://syzkaller.appspot.com/bug?extid=88412ee8811832b00dbe > compiler: clang version 14.0.0 (git@github.com:llvm/llvm-project.git 0996585c8e3b3d409494eb5f1cad714b9e1f7fb5), GNU ld (GNU Binutils for Debian) 2.35.2 > syz repro: https://syzkaller.appspot.com/x/repro.syz?x=13a7abf9b00000 > C reproducer: https://syzkaller.appspot.com/x/repro.c?x=172410b9b00000 > > IMPORTANT: if you fix the issue, please add the following tag to the commit: > Reported-by: syzbot+88412ee8811832b00dbe(a)syzkaller.appspotmail.com > > ===================================================== > BUG: KMSAN: uninit-value in snd_rawmidi_ioctl+0xf1e/0x1330 sound/core/rawmidi.c:887 > snd_rawmidi_ioctl+0xf1e/0x1330 sound/core/rawmidi.c:887 Nice catch. I wonder why the problem was not detected by tests. The patch bellow should initialize the user_pversion variable: diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6f30231bdb88..befa9809ff00 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -447,6 +447,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) err = -ENOMEM; goto __error; } + rawmidi_file->user_pversion = 0; init_waitqueue_entry(&wait, current); add_wait_queue(&rmidi->open_wait, &wait); while (1) { I'll send the path in the proper format ASAP. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

ALSA update from 1.2.5.1 to 1.2.6.1 loses audio devices on Dell XPS 9700
by GitHub issues - opened 18 Dec '21

18 Dec '21

alsa-project/alsa-lib issue #206 was opened from roland-d: Hello, On December 16, Manjaro pushed an update which included an ALSA update from 1.2.5.1 to 1.2.6.1. This has caused users of a DELL XPS 9700 to no longer show any audio devices except the built-in speaker. This has been reported in the release post as well by a user named sparcplug. https://forum.manjaro.org/t/stable-update-2021-12-16-kernels-kde-gear-21-12… I can confirm that for me the audio devices also disappeared. A rollback to 1.2.5.1 and restart of the system brought back the sound devices. I have no idea what causes this but wanted to report it and if you need me to test I am happy to do so. Issue URL : https://github.com/alsa-project/alsa-lib/issues/206 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

rawmidi driver dropping some sysex commands
by Kolten Pearson 17 Dec '21

17 Dec '21

I have been working on reverse engineering the sysex commands to control an Arturia MiniLab mkII controller, and have run into an issue where the rawmidi driver is dropping some of the commands. Here is a snippet of my code that illustrates the problem --- snd_rawmidi_t* midi_input; snd_rawmidi_t* midi_output; void midi_init() { int ret = snd_rawmidi_open(&midi_input, &midi_output, "hw:3,0,0", SND_RAWMIDI_NONBLOCK); assert (ret == 0 && "could not open midi device"); uint8_t pad_list[] = { MA_PAD_1, MPC_NO_COLOR, MA_PAD_2, MPC_RED, MA_PAD_3, MPC_GREEN, MA_PAD_4, MPC_YELLOW, MA_PAD_5, MPC_BLUE, MA_PAD_6, MPC_CYAN, MA_PAD_7, MPC_MAGENTA, MA_PAD_8, MPC_WHITE, MA_PAD_9, MPC_NO_COLOR, MA_PAD_10, MPC_RED, MA_PAD_11, MPC_GREEN, MA_PAD_12, MPC_YELLOW, MA_PAD_13, MPC_BLUE, MA_PAD_14, MPC_CYAN, MA_PAD_15, MPC_MAGENTA, MA_PAD_16, MPC_WHITE, }; for (int i = 0; i < 16; i++) { midi_set(MC_ON_OFF, pad_list[2*i], 0x08); //sets the pad to be a control instead of a note midi_set(MC_MIN_RANGE, pad_list[2*i], 0); midi_set(MC_MAX_RANGE, pad_list[2*i], 1); midi_set(MC_MODE_SET, pad_list[2*i], 1); //set it to gate, 0 is toggle midi_set(MC_PAD_COLOR, pad_list[2*i], pad_list[2*i+1]); } midi_set(MC_GLOBAL_SET, MA_PAD_BACKLIGHTS, 0x0f); //updates the pad backlights uint8_t knob_list[] = { MA_KNOB_1, MA_KNOB_2, MA_KNOB_3, MA_KNOB_4, MA_KNOB_5, MA_KNOB_6, MA_KNOB_7, MA_KNOB_8, MA_KNOB_9, MA_KNOB_10, MA_KNOB_11, MA_KNOB_12, MA_KNOB_13, MA_KNOB_14, MA_KNOB_15, MA_KNOB_16 }; for (int i = 0; i < 16; i++) { midi_set(MC_MODE_SET, knob_list[i], 0x02); //sets it to relative mode where 0-3 is right turn, 7f-7d is left turn midi_set(MC_CC_SET, knob_list[i], i); } } void midi_set(enum midi_set_command command, enum midi_set_address address, uint8_t data) { uint8_t raw_command[] = {0xf0, 0x00, 0x20, 0x6b, 0x7f, 0x42, 0x02, 0x00, command, address, data, 0xf7}; int count = snd_rawmidi_write(midi_output, raw_command, sizeof(raw_command)); assert(count == sizeof(raw_command) && "bad write"); //SDL_Delay(1); int ret = snd_rawmidi_drain(midi_output); assert(ret == 0 && "could not drain"); } --- When this code is run, the minilab will only respond to a nondeterministic number of calls to the midi_set function and ignore the rest. However if the SDL_Delay(1) line is uncommented, it will work correctly every time (This function call causes the process to sleep for at least 1 millisecond). I feel like the call to snd_rawmidi_drain should make the SDL_Delay call unnecessary, but that is not the case. Is there something I am not understanding about how the api works? Thanks for any help!

2 1

ALSA 1.2.6.3 release
by Jaroslav Kysela 17 Dec '21

17 Dec '21

Hello all, new ALSA userspace packages were released. This is third bugfix release. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-ucm-conf Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.6.2_v1.2.6.3 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[PATCH v5 00/11] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 17 Dec '21

17 Dec '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. (!) These patches are made on top of stable dmaengine branch from Vinod Koul. The following changes since commit fa55b7dcdc43c1aa1ba12bca9d2dd4318c2a0dbf: Linux 5.16-rc1 (2021-11-14 13:56:52 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/dmaengine.git tags/dmaengine_topic_slave_id_removal_5.17 for you to fetch changes up to 3c219644075795a99271d345efdfa8b256e55161: dmaengine: remove slave_id config field (2021-12-17 11:23:56 +0530) Changelog: v5: - Dropped all patches that were already applied by Thierry Reding. - Made minor changes that were suggested by Thierry Reding. - Added acks from Thierry Reding. - Rebased patches on top of stable dmaengine branch from Vinod Koul. v4: - Added patches that update multi_v7_defconfig with the enabled S/PDIF and APB DMA drivers. v3: - Renamed S/PDIF device-tree clocks as was suggested by Rob Herring. - Added r-bs and acks that were given by Rob Herring to v2. v2: - Corrected I2S yaml problem that was reported by the DT bot for v1 by removing the non-existent required clock-names property. - Removed assigned-clocks property from S/PDIF yaml since this property is now inherited from the clocks property. - Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making it the first sound/soc patch in the series, like it was suggested by Mark Brown in the comment to v1. Also reworded commit message of this patch to *not* make it looks like it should be backported to stable kernels. Dmitry Osipenko (11): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 77 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 85 ++++++++ sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 198 +++++++++++++----- sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 8 files changed, 359 insertions(+), 88 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

1 12

[PATCH 0/6] ASoC : soc-pcm: fix trigger race conditions with shared BE
by Pierre-Louis Bossart 17 Dec '21

17 Dec '21

We've been adding a 'deep buffer' PCM device to several SOF topologies in order to reduce power consumption. The typical use-case would be music playback over a headset: this additional PCM device provides more buffering and longer latencies, leaving the rest of the system sleep for longer periods. Notifications and 'regular' low-latency audio playback would still use the 'normal' PCM device and be mixed with the 'deep buffer' before rendering on the headphone endpoint. The tentative direction would be to expose this alternate device to PulseAudio/PipeWire/CRAS via the UCM SectionModifier definitions. That seemed a straightforward topology change until our automated validation stress tests started reporting issues on SoundWire platforms, when e.g. two START triggers might be send and conversely the STOP trigger is never sent. The SoundWire stream state management flagged inconsistent states when the two 'normal' and 'deep buffer' devices are used concurrently with rapid play/stop/pause monkey testing. Looking at the soc-pcm.c code, it seems that the BE state management needs a lot of love. a) there is no consistent protection for the BE state. In some parts of the code, the state updates are protected by a spinlock but in the trigger they are not. When we open/play/close the two PCM devices in stress tests, we end-up testing a state that is being modified. That can't be good. b) there is a conceptual deadlock: on stop we check the FE states to see if a shared BE can be stopped, but since we trigger the BE first the FE states have not been modified yet, so the TRIGGER_STOP is never sent. This patchset suggests the removal of the dedicated 'dpcm_lock' and follows the design suggested by Takashi Iwai. By default the protection relies on the 'pcm_mutex', except for the FE and BE triggers where the mutex cannot be used. In this case, the FE PCM lock is used instead. In the cases where a BE is added/removed, the pcm_mutex and FE PCM lock are both taken. In addition, the BE PCM lock is used to serialize access to a shared BE. With these patches I am able to run our entire validation suite without any issues with this new 'deep buffer' topology, and no regressions on existing solutions [1]. The tests were reproduced by Bard Liao for SoundWire devices. One might ask 'how come we didn't see this earlier'? The answer is probably that the .trigger callbacks in most implementations seems to perform DAPM operations, and sending the triggers multiple times is not an issue. In the case of SoundWire, we do use the .trigger callback to reconfigure the bus using the 'bank switch' mechanism. It could be acceptable to tolerate a trigger multiple times, but the deadlock on stop cannot be fixed at the SoundWire level alone. Opens: 1) The issues reported by Nvidia on the RFCv3 may or may not be present. We'd need test results to make sure the locking update does not introduce a regression on Tegra. 2) There are other reports of kernel oopses [2] that seem related to the lack of protection. I'd be good to confirm if this patchset solve these problems as well. [1] https://github.com/thesofproject/linux/pull/3146 [2] https://lore.kernel.org/alsa-devel/002f01d7b4f5$c030f4a0$4092dde0$@samsung.… changes since RFCv3: Used two patches from Takashi. We now use the pcm_mutex, the FE stream lock when adding and deleting a BE, and the BE stream lock to handle concurrency between streams using the same BE. Added a patch to use GFP_ATOMIC for the DPCM structure. Fixed PAUSE_RELEASE transition (GitHub comment from Kai Vehmanen) changes since RFCv2: Removal of dpcm_lock to use FE PCM locks (credits to Takashi Iwai for the suggestion). The FE PCM lock is now used before each use of for_each_dpcm_be() - with the exception of the trigger where the lock is already taken. This change is also applied in drivers which make use of this loop (compress, SH, FSL). Addition of BE PCM lock to deal with mutual exclusion between triggers for the same BE. Alignment of the BE atomicity on the FE on connections, this is required to avoid sleeping in atomic context. Additional cleanups (indentation, static functions) changes since RFC v1: Removed unused function Removed exported symbols only used in soc-pcm.c, used static instead Use a mutex instead of a spinlock Protect all for_each_dpcm_be() loops Fix bugs introduced in the refcount Pierre-Louis Bossart (4): ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure ASoC: soc-pcm: align BE 'atomicity' with that of the FE ASoC: soc-pcm: test refcount before triggering ASoC: soc-pcm: fix BE handling of PAUSE_RELEASE Takashi Iwai (2): ASoC: soc-pcm: Fix and cleanup DPCM locking ASoC: soc-pcm: serialize BE triggers include/sound/soc-dpcm.h | 2 + include/sound/soc.h | 2 - sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 351 +++++++++++++++++++++++++++------------ 4 files changed, 246 insertions(+), 110 deletions(-) -- 2.25.1

3 11

[PATCH] ASoC: max98357a: Add mixer control to mute/unmute speaker
by Ajit Kumar Pandey 17 Dec '21

17 Dec '21

Add "Playback Switch" mixer control to mute or unmute speaker playback from ucm conf depend on use cases. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/codecs/max98357a.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 918812763884..9b2f16ab4498 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -73,6 +73,36 @@ static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, return 0; } +static int speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = max98357a->sdmode_switch; + + return 0; +} + +static int speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + int mode = ucontrol->value.enumerated.item[0]; + + max98357a->sdmode_switch = mode; + gpiod_set_value_cansleep(max98357a->sdmode, mode); + dev_dbg(component->dev, "set sdmode to %d", mode); + + return 0; +} + +static const struct snd_kcontrol_new max98357a_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Playback Switch", 0, + speaker_mute_get, speaker_mute_put), +}; + static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Speaker"), SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0, @@ -86,6 +116,8 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { }; static const struct snd_soc_component_driver max98357a_component_driver = { + .controls = max98357a_snd_controls, + .num_controls = ARRAY_SIZE(max98357a_snd_controls), .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, -- 2.25.1

3 7

[PATCH v5 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 17 Dec '21

17 Dec '21

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 27 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 144 ++++ sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2051 insertions(+), 1092 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

2 11

[PATCH] ASoC: wm8962: potential use of error pointer
by Jiasheng Jiang 17 Dec '21

17 Dec '21

The return value of devm_clk_get() needs to be checked. To avoid use of error pointer in case of the failure of alloc. Fixes: d74e9e7090ae ("ASoC: wm8962: Add device tree binding") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 34080f497584..66693bf17763 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3540,6 +3540,8 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, } pdata->mclk = devm_clk_get(&i2c->dev, NULL); + if (IS_ERR(pdata->mclk)) + return -ENOMEM; return 0; } -- 2.25.1

3 2

[PATCH 1/2] ASoC: mediatek: mt8195: update control for RT5682 series
by Trevor Wu 17 Dec '21

17 Dec '21

Playback pop is observed and the root cause is the reference clock provided by MT8195 is diabled before RT5682 finishes the control flow. To ensure the reference clock supplied to RT5682 is disabled after RT5682 finishes all register controls. We replace BCLK with MCLK for RT5682 reference clock, and makes use of set_bias_level_post to handle MCLK which guarantees MCLK is off after all RT5682 register access. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 66 +++++++++++++++- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 76 ++++++++++++++++--- 2 files changed, 128 insertions(+), 14 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c index 5cdbfaafd479..05359365f200 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c @@ -37,6 +37,7 @@ struct mt8195_mt6359_rt1011_rt5682_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; + struct clk *i2so1_mclk; }; static const struct snd_soc_dapm_widget @@ -87,8 +88,8 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_BCLK1, - rate * 64, rate * 512); + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_MCLK, + rate * 256, rate * 512); if (ret) { dev_err(card->dev, "failed to set pll\n"); return ret; @@ -101,7 +102,7 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 128, + return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 256, SND_SOC_CLOCK_OUT); } @@ -565,6 +566,51 @@ static const struct snd_soc_ops mt8195_capture_ops = { .startup = mt8195_capture_startup, }; +static int mt8195_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct mt8195_mt6359_rt1011_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * It's required to control mclk directly in the set_bias_level_post + * function for rt5682 and rt5682s codec, or the unexpected pop happens + * at the end of playback. + */ + if (!component || + (strcmp(component->name, RT5682_DEV0_NAME) && + strcmp(component->name, RT5682S_DEV0_NAME))) + return 0; + + if (IS_ERR(priv->i2so1_mclk)) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (!__clk_is_enabled(priv->i2so1_mclk)) + return 0; + + dev_dbg(card->dev, "Disable i2so1"); + clk_disable_unprepare(priv->i2so1_mclk); + break; + case SND_SOC_BIAS_ON: + dev_dbg(card->dev, "Enable i2so1"); + ret = clk_prepare_enable(priv->i2so1_mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + break; + default: + break; + } + + return ret; +} + + enum { DAI_LINK_DL2_FE, DAI_LINK_DL3_FE, @@ -1040,6 +1086,7 @@ static struct snd_soc_card mt8195_mt6359_rt1011_rt5682_soc_card = { .num_dapm_routes = ARRAY_SIZE(mt8195_mt6359_rt1011_rt5682_routes), .codec_conf = rt1011_amp_conf, .num_configs = ARRAY_SIZE(rt1011_amp_conf), + .set_bias_level_post = mt8195_set_bias_level_post, }; static int mt8195_mt6359_rt1011_rt5682_dev_probe(struct platform_device *pdev) @@ -1072,6 +1119,19 @@ static int mt8195_mt6359_rt1011_rt5682_dev_probe(struct platform_device *pdev) return -EINVAL; } + priv->i2so1_mclk = devm_clk_get(&pdev->dev, "i2so1_mclk"); + if (IS_ERR(priv->i2so1_mclk)) { + ret = PTR_ERR(priv->i2so1_mclk); + if (ret == -ENOENT) { + dev_dbg(&pdev->dev, + "Failed to get i2so1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get i2so1_mclk, err:%d\n", ret); + return ret; + } + for_each_card_prelinks(card, i, dai_link) { if (!dai_link->platforms->name) dai_link->platforms->of_node = priv->platform_node; diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index fa50a31e9718..b2c3b57da9c4 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -50,6 +50,7 @@ struct mt8195_mt6359_rt1019_rt5682_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; + struct clk *i2so1_mclk; }; static const struct snd_soc_dapm_widget @@ -96,8 +97,6 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int rate = params_rate(params); - unsigned int mclk_fs_ratio = 128; - unsigned int mclk_fs = rate * mclk_fs_ratio; int bitwidth; int ret; @@ -113,25 +112,22 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, - RT5682_PLL1_S_BCLK1, - params_rate(params) * 64, - params_rate(params) * 512); + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_MCLK, + rate * 256, rate * 512); if (ret) { dev_err(card->dev, "failed to set pll\n"); return ret; } - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5682_SCLK_S_PLL1, - params_rate(params) * 512, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + rate * 512, SND_SOC_CLOCK_IN); if (ret) { dev_err(card->dev, "failed to set sysclk\n"); return ret; } - return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_fs, SND_SOC_CLOCK_OUT); + return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 256, + SND_SOC_CLOCK_OUT); } static const struct snd_soc_ops mt8195_rt5682_etdm_ops = { @@ -564,6 +560,50 @@ static const struct snd_soc_ops mt8195_capture_ops = { .startup = mt8195_capture_startup, }; +static int mt8195_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct mt8195_mt6359_rt1019_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * It's required to control mclk directly in the set_bias_level_post + * function for rt5682 and rt5682s codec, or the unexpected pop happens + * at the end of playback. + */ + if (!component || + (strcmp(component->name, RT5682_DEV0_NAME) && + strcmp(component->name, RT5682S_DEV0_NAME))) + return 0; + + if (IS_ERR(priv->i2so1_mclk)) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (!__clk_is_enabled(priv->i2so1_mclk)) + return 0; + + dev_dbg(card->dev, "Disable i2so1"); + clk_disable_unprepare(priv->i2so1_mclk); + break; + case SND_SOC_BIAS_ON: + dev_dbg(card->dev, "Enable i2so1"); + ret = clk_prepare_enable(priv->i2so1_mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + break; + default: + break; + } + + return ret; +} + enum { DAI_LINK_DL2_FE, DAI_LINK_DL3_FE, @@ -1203,6 +1243,7 @@ static struct snd_soc_card mt8195_mt6359_rt1019_rt5682_soc_card = { .num_dapm_widgets = ARRAY_SIZE(mt8195_mt6359_rt1019_rt5682_widgets), .dapm_routes = mt8195_mt6359_rt1019_rt5682_routes, .num_dapm_routes = ARRAY_SIZE(mt8195_mt6359_rt1019_rt5682_routes), + .set_bias_level_post = mt8195_set_bias_level_post, }; static int mt8195_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, @@ -1285,6 +1326,19 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) return -EINVAL; } + priv->i2so1_mclk = devm_clk_get(&pdev->dev, "i2so1_mclk"); + if (IS_ERR(priv->i2so1_mclk)) { + ret = PTR_ERR(priv->i2so1_mclk); + if (ret == -ENOENT) { + dev_dbg(&pdev->dev, + "Failed to get i2so1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get i2so1_mclk, err:%d\n", ret); + return ret; + } + /* dai link */ priv->adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); -- 2.18.0

3 6

Re: [PATCH v2 1/1] ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output
by Ajit Kumar Pandey 17 Dec '21

17 Dec '21

On 12/9/2021 7:55 AM, Shuming [范書銘] wrote: > [CAUTION: External Email] > >> Subject: [PATCH v2 1/1] ASoC: rt5682s: Add dapm switch to mute/unmute HP >> playback output >> >> Add dapm switch in playback path to mute or unmute HP output data. >> We will set and reset MUTE_SFT bit in RT5682S_HP_CTRL_1 register based on >> switch value to mute or unmute respective channel. >> >> Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> >> --- >> sound/soc/codecs/rt5682s.c | 19 +++++++++++++++++-- >> 1 file changed, 17 insertions(+), 2 deletions(-) >> >> diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index >> d49a4f68566d..21ab5f7df422 100644 >> --- a/sound/soc/codecs/rt5682s.c >> +++ b/sound/soc/codecs/rt5682s.c >> @@ -1573,6 +1573,14 @@ static const char * const >> rt5682s_adcdat_pin_select[] = { >> "ADCDAT1", "ADCDAT2", >> }; >> >> +/* Out Switch */ >> +static const struct snd_kcontrol_new hpol_switch = >> + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, >> + RT5682S_L_MUTE_SFT, 1, 1); >> +static const struct snd_kcontrol_new hpor_switch = >> + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, >> + RT5682S_R_MUTE_SFT, 1, 1); >> + >> static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_adcdat_pin_enum, >> RT5682S_GPIO_CTRL_1, RT5682S_GP4_PIN_SFT, >> RT5682S_GP4_PIN_MASK, >> rt5682s_adcdat_pin_select, rt5682s_adcdat_pin_values); @@ -1746,6 >> +1754,11 @@ static const struct snd_soc_dapm_widget >> rt5682s_dapm_widgets[] = { >> SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, >> rt5682s_hp_amp_event, >> SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), >> >> + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, >> + &hpol_switch), >> + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, >> + &hpor_switch), >> + >> /* CLK DET */ >> SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, >> RT5682S_SYS_CLK_DET_SFT, 0, NULL, 0), @@ -1895,8 +1908,10 @@ >> static const struct snd_soc_dapm_route rt5682s_dapm_routes[] = { >> {"HP Amp", NULL, "CLKDET SYS"}, >> {"HP Amp", NULL, "SAR"}, >> >> - {"HPOL", NULL, "HP Amp"}, >> - {"HPOR", NULL, "HP Amp"}, >> + {"HPOL Playback", "Switch", "HP Amp"}, >> + {"HPOR Playback", "Switch", "HP Amp"}, >> + {"HPOL", NULL, "HPOL Playback"}, >> + {"HPOR", NULL, "HPOR Playback"}, >> }; > > RT5682S enables the 1 bit control for HP output. > Therefore, the setting of RT5682S_HP_CTRL_1[15][7] will be not effective. > > Ok, it was actually effective and muted audio when we test in our platform, possible because i'm tearing down dapm path based on switch value. If we don't have register bits available to mute i'll resubmit the change with virtual switch widget to disable DAPM and stop audio.

2 1

Question about soc_pcm_apply_msb()
by Kuninori Morimoto 17 Dec '21

17 Dec '21

Hi ALSA ML At soc_pcm_apply_msb(), (A) part finds max sig_bits for Codec, and (B) part finds max sig_bits for CPU and, set it to substream via soc_pcm_set_msb() at (X), (Y). static void soc_pcm_apply_msb() { ... ^ for_each_rtd_codec_dais(rtd, i, codec_dai) { (A) ... v bits = max(pcm_codec->sig_bits, bits); } ^ for_each_rtd_cpu_dais(rtd, i, cpu_dai) { (B) ... v cpu_bits = max(pcm_cpu->sig_bits, cpu_bits); } (X) soc_pcm_set_msb(substream, bits); (Y) soc_pcm_set_msb(substream, cpu_bits); } I wonder do we need both (X) (Y) ? I think we can merge (A) and (B) (= find Codec/CPU max sig_bits), and call soc_pcm_set_msb() once, but am I misunderstand ? We have many patch around here, and below are the main patches. The 1st patch has both (X)(Y) code. new 19bdcc7aeed4169820be6a683c422fc06d030136 ("ASoC: Add multiple CPU DAI support for PCM ops") 57be92066f68e63bd4a72a65d45c3407c0cb552a ("ASoC: soc-pcm: cleanup soc_pcm_apply_msb()") c8dd1fec47d0b1875f292c40bed381b343e38b40 ("ASoC: pcm: Refactor soc_pcm_apply_msb for multicodecs") 58ba9b25454fe9b6ded804f69cb7ed4500b685fc ("ASoC: Allow drivers to specify how many bits are significant on a DAI") old Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[RESEND, PATCH v7 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 16 Dec '21

16 Dec '21

DT bindings for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..83936f594d1a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: audio-codec@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..ec988439da20 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH] dt-bindings: dsp: mediatek: add mt8195 dsp document
by YC Hung 16 Dec '21

16 Dec '21

From: "yc.hung" <yc.hung(a)mediatek.com> This patch adds mt8195 dsp document. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- .../bindings/dsp/mtk,mt8195-dsp.yaml | 116 ++++++++++++++++++ 1 file changed, 116 insertions(+) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml diff --git a/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml new file mode 100644 index 000000000000..33db11ee2336 --- /dev/null +++ b/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/dsp/mtk,mt8195-dsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek mt8195 DSP core + +maintainers: + - YC Hung <yc.hung(a)mediatek.com> + +description: | + Some boards from mt8195 contain a DSP core used for + advanced pre- and post- audio processing. +properties: + compatible: + const: mediatek,mt8195-dsp + + reg: + items: + - description: Address and size of the DSP Cfg registers + - description: Address and size of the DSP SRAM + + reg-names: + items: + - const: cfg + - const: sram + + interrupts: + items: + - description: watchdog interrupt + + interrupt-names: + items: + - const: wdt + + clocks: + items: + - description: mux for audio dsp clock + - description: 26M clock + - description: mux for audio dsp local bus + - description: default audio dsp local bus clock source + - description: clock gate for audio dsp clock + - description: mux for audio dsp access external bus + + clock-names: + items: + - const: adsp_sel + - const: clk26m_ck + - const: audio_local_bus + - const: mainpll_d7_d2 + - const: scp_adsp_audiodsp + - const: audio_h + + power-domains: + maxItems: 1 + + mboxes: + items: + - description: a mailbox is ised for ipc reply between host and audio DSP. + - description: a mailbox is ised for ipc reuqest between host and audio DSP. + + mbox-names: + items: + - const: mbox0 + - const: mbox1 + + memory-region: + items: + - description: A phandle to a reserved memory region is used for dma buffer between host and DSP. + - description: A phandle to a reserved memory region is used for DSP system memory. + +required: + - compatible + - reg + - reg-names + - interrupts + - interrupt-names + - clocks + - clock-names + - memory-region + - power-domains + - mbox-names + - mboxes + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + adsp: dsp@10803000 { + compatible = "mediatek,mt8195-dsp"; + reg = <0x10803000 0x1000>, + <0x10840000 0x40000>; + reg-names = "cfg", "sram"; + interrupts = <GIC_SPI 694 IRQ_TYPE_LEVEL_HIGH 0>; + interrupt-names = "wdt"; + clocks = <&topckgen 10>, //CLK_TOP_ADSP + <&clk26m>, + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 136>, //CLK_TOP_MAINPLL_D7_D2 + <&scp_adsp 0>, //CLK_SCP_ADSP_AUDIODSP + <&topckgen 34>; //CLK_TOP_AUDIO_H + clock-names = "adsp_sel", + "clk26m_ck", + "audio_local_bus", + "mainpll_d7_d2", + "scp_adsp_audiodsp", + "audio_h"; + memory-region = <&adsp_dma_mem_reserved>, + <&adsp_mem_reserved>; + power-domains = <&spm 6>; //MT8195_POWER_DOMAIN_ADSP + mbox-names = "mbox0", "mbox1"; + mboxes = <&adsp_mailbox0>, <&adsp_mailbox1>; + }; -- 2.18.0

2 1

[PATCH v2] dt-bindings: sound: nvidia, tegra-audio: Convert multiple txt bindings to yaml
by David Heidelberg 16 Dec '21

16 Dec '21

Convert Tegra audio complex with the * ALC5632 * MAX98090 * RT5640 * RT5677 * SGTL5000 * TrimSlice * WM8753 * WM8903 * WM9712 codec to the YAML format. Additional changes: - added missing HPOUTL to the WM9712 codec. - extended rt5677 codec with multiple pins Reviewed-by: Dmitry Osipenko <digetx(a)gmail.com> Signed-off-by: David Heidelberg <david(a)ixit.cz> --- v2: - converted from one big schema into the common properties and then references to it - few additional changes mentioned in commit message .../sound/nvidia,tegra-audio-alc5632.txt | 48 -------- .../sound/nvidia,tegra-audio-alc5632.yaml | 74 +++++++++++++ .../sound/nvidia,tegra-audio-common.yaml | 83 ++++++++++++++ .../sound/nvidia,tegra-audio-max98090.txt | 53 --------- .../sound/nvidia,tegra-audio-max98090.yaml | 97 +++++++++++++++++ .../sound/nvidia,tegra-audio-rt5640.txt | 52 --------- .../sound/nvidia,tegra-audio-rt5640.yaml | 85 +++++++++++++++ .../sound/nvidia,tegra-audio-rt5677.txt | 67 ------------ .../sound/nvidia,tegra-audio-rt5677.yaml | 103 ++++++++++++++++++ .../sound/nvidia,tegra-audio-sgtl5000.txt | 42 ------- .../sound/nvidia,tegra-audio-sgtl5000.yaml | 67 ++++++++++++ .../sound/nvidia,tegra-audio-trimslice.txt | 21 ---- .../sound/nvidia,tegra-audio-trimslice.yaml | 33 ++++++ .../sound/nvidia,tegra-audio-wm8753.txt | 40 ------- .../sound/nvidia,tegra-audio-wm8753.yaml | 79 ++++++++++++++ .../sound/nvidia,tegra-audio-wm8903.txt | 62 ----------- .../sound/nvidia,tegra-audio-wm8903.yaml | 93 ++++++++++++++++ .../sound/nvidia,tegra-audio-wm9712.txt | 60 ---------- .../sound/nvidia,tegra-audio-wm9712.yaml | 76 +++++++++++++ 19 files changed, 790 insertions(+), 445 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt deleted file mode 100644 index 57f40f93453e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt +++ /dev/null @@ -1,48 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-alc5632" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the ALC5632's pins as documented in the binding for the device - and: - - * Headset Stereophone - * Int Spk - * Headset Mic - * Digital Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller -- nvidia,audio-codec : The phandle of the ALC5632 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-alc5632-paz00", - "nvidia,tegra-audio-alc5632"; - - nvidia,model = "Compal PAZ00"; - - nvidia,audio-routing = - "Int Spk", "SPK_OUTP", - "Int Spk", "SPK_OUTN", - "Headset Mic","MICBIAS1", - "MIC1_N", "Headset Mic", - "MIC1_P", "Headset Mic", - "Headset Stereophone", "HP_OUT_R", - "Headset Stereophone", "HP_OUT_L"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&alc5632>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml new file mode 100644 index 000000000000..7ef774910e5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-alc5632.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with ALC5632 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-alc5632(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-alc5632 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headset Stereophone" + - "Int Spk" + - "Headset Mic" + - "Digital Mic" + + # CODEC Pins + - SPKOUT + - SPKOUTN + - MICBIAS1 + - MIC1 + - HPR + - HPL + - DMICDAT + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = "Int Spk", "SPKOUT", + "Int Spk", "SPKOUTN", + "Headset Mic", "MICBIAS1", + "MIC1", "Headset Mic", + "Headset Stereophone", "HPR", + "Headset Stereophone", "HPL", + "DMICDAT", "Digital Mic"; + + nvidia,i2s-controller = <&i2s>; + nvidia,audio-codec = <&codec>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml new file mode 100644 index 000000000000..82801b4f46dd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-common.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/nvidia,tegra-audio-common.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Common properties for NVIDIA Tegra audio complexes + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +properties: + clocks: + items: + - description: PLL A clock + - description: PLL A OUT0 clock + - description: The Tegra cdev1/extern1 clock, which feeds the card's mclk + + clock-names: + items: + - const: pll_a + - const: pll_a_out0 + - const: mclk + + nvidia,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + + nvidia,ac97-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the AC97 controller + + nvidia,i2s-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the Tegra I2S controller + + nvidia,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of audio codec + + nvidia,spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + nvidia,hp-mute-gpios: + maxItems: 1 + description: The GPIO that mutes the headphones + + nvidia,hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + nvidia,mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphone is plugged in + + nvidia,ear-sel-gpios: + maxItems: 1 + description: The GPIO that switch between the microphones + + nvidia,int-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the internal microphone + + nvidia,ext-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the external microphone + + nvidia,headset: + type: boolean + description: The Mic Jack represents state of the headset microphone pin + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt deleted file mode 100644 index c3495beba358..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ /dev/null @@ -1,53 +0,0 @@ -NVIDIA Tegra audio complex, with MAX98090 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-max98090" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the MAX98090's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - * Int Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the MAX98090 audio codec. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-max98090-venice2", - "nvidia,tegra-audio-max98090"; - nvidia,model = "NVIDIA Tegra Venice2"; - - nvidia,audio-routing = - "Headphones", "HPR", - "Headphones", "HPL", - "Speakers", "SPKR", - "Speakers", "SPKL", - "Mic Jack", "MICBIAS", - "IN34", "Mic Jack"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&acodec>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml new file mode 100644 index 000000000000..ccc2ee77ca30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.yaml @@ -0,0 +1,97 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with MAX98090 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98090(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-max98090 + - items: + - enum: + - nvidia,tegra-audio-max98090-nyan-big + - nvidia,tegra-audio-max98090-nyan-blaze + - const: nvidia,tegra-audio-max98090-nyan + - const: nvidia,tegra-audio-max98090 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphones" + - "Speakers" + - "Mic Jack" + - "Int Mic" + + # CODEC Pins + - MIC1 + - MIC2 + - DMICL + - DMICR + - IN1 + - IN2 + - IN3 + - IN4 + - IN5 + - IN6 + - IN12 + - IN34 + - IN56 + - HPL + - HPR + - SPKL + - SPKR + - RCVL + - RCVR + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra124-car.h> + + sound { + compatible = "nvidia,tegra-audio-max98090-venice2", + "nvidia,tegra-audio-max98090"; + nvidia,model = "NVIDIA Tegra Venice2"; + + nvidia,audio-routing = + "Headphones", "HPR", + "Headphones", "HPL", + "Speakers", "SPKR", + "Speakers", "SPKL", + "Mic Jack", "MICBIAS", + "IN34", "Mic Jack"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&acodec>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt deleted file mode 100644 index 7788808dcd0b..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ /dev/null @@ -1,52 +0,0 @@ -NVIDIA Tegra audio complex, with RT5640 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5640" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the RT5640's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5640 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5640-dalmore", - "nvidia,tegra-audio-rt5640"; - nvidia,model = "NVIDIA Tegra Dalmore"; - - nvidia,audio-routing = - "Headphones", "HPOR", - "Headphones", "HPOL", - "Speakers", "SPORP", - "Speakers", "SPORN", - "Speakers", "SPOLP", - "Speakers", "SPOLN"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5640>; - - nvidia,hp-det-gpios = <&gpio 143 0>; /* GPIO PR7 */ - - clocks = <&tegra_car 216>, <&tegra_car 217>, <&tegra_car 120>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml new file mode 100644 index 000000000000..e768fb0e9a59 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5640.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5639 or RT5640 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt56(39|40)(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5640 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphones" + - "Speakers" + - "Mic Jack" + + # CODEC Pins + - DMIC1 + - DMIC2 + - MICBIAS1 + - IN1P + - IN1R + - IN2P + - IN2R + - HPOL + - HPOR + - LOUTL + - LOUTR + - MONOP + - MONON + - SPOLP + - SPOLN + - SPORP + - SPORN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5640-dalmore", + "nvidia,tegra-audio-rt5640"; + nvidia,model = "NVIDIA Tegra Dalmore"; + + nvidia,audio-routing = + "Headphones", "HPOR", + "Headphones", "HPOL", + "Speakers", "SPORP", + "Speakers", "SPORN", + "Speakers", "SPOLP", + "Speakers", "SPOLN"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5640>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + + clocks = <&clk 216>, <&clk 217>, <&clk 120>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; + diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt deleted file mode 100644 index a4589cda214e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt +++ /dev/null @@ -1,67 +0,0 @@ -NVIDIA Tegra audio complex, with RT5677 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5677" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the RT5677's pins (as documented in its binding), and the jacks - on the board: - - * Headphone - * Speaker - * Headset Mic - * Internal Mic 1 - * Internal Mic 2 - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in -- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier -- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in -- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5677-ryu", - "nvidia,tegra-audio-rt5677"; - nvidia,model = "NVIDIA Tegra Ryu"; - - nvidia,audio-routing = - "Headphone", "LOUT2", - "Headphone", "LOUT1", - "Headset Mic", "MICBIAS1", - "IN1P", "Headset Mic", - "IN1N", "Headset Mic", - "DMIC L1", "Internal Mic 1", - "DMIC R1", "Internal Mic 1", - "DMIC L2", "Internal Mic 2", - "DMIC R2", "Internal Mic 2", - "Speaker", "PDM1L", - "Speaker", "PDM1R"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5677>; - - nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; - nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; - nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; - nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml new file mode 100644 index 000000000000..03ff691c26c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-rt5677.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with RT5677 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-rt5677(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-rt5677 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone" + - "Speaker" + - "Headset Mic" + - "Internal Mic 1" + - "Internal Mic 2" + + # CODEC Pins + - IN1P + - IN1N + - IN2P + - IN2N + - MICBIAS1 + - DMIC1 + - DMIC2 + - DMIC3 + - DMIC4 + - "DMIC L1" + - "DMIC L2" + - "DMIC L3" + - "DMIC L4" + - "DMIC R1" + - "DMIC R2" + - "DMIC R3" + - "DMIC R4" + - LOUT1 + - LOUT2 + - LOUT3 + - PDM1L + - PDM1R + - PDM2L + - PDM2R + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio 143 0>; + nvidia,mic-present-gpios = <&gpio 132 1>; + nvidia,hp-en-gpios = <&rt5677 1 0>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 1>; + + clocks = <&clk 216>, + <&clk 217>, + <&clk 121>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt deleted file mode 100644 index 5da7da4ea07a..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt +++ /dev/null @@ -1,42 +0,0 @@ -NVIDIA Tegra audio complex, with SGTL5000 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-sgtl5000" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the SGTL5000's pins (as documented in its binding), and the jacks - on the board: - - * Headphone Jack - * Line In Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the SGTL5000 audio codec. - -Example: - -sound { - compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", - "nvidia,tegra-audio-sgtl5000"; - nvidia,model = "Toradex Apalis T30"; - nvidia,audio-routing = - "Headphone Jack", "HP_OUT", - "LINE_IN", "Line In Jack", - "MIC_IN", "Mic Jack"; - nvidia,i2s-controller = <&tegra_i2s2>; - nvidia,audio-codec = <&sgtl5000>; - clocks = <&tegra_car TEGRA30_CLK_PLL_A>, - <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA30_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml new file mode 100644 index 000000000000..943e7c01741c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with SGTL5000 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-sgtl5000([-_][a-z0-9]+)+$' + - const: nvidia,tegra-audio-sgtl5000 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Line In Jack" + - "Mic Jack" + + # CODEC Pins + - HP_OUT + - LINE_OUT + - LINE_IN + - MIC_IN + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + sound { + compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", + "nvidia,tegra-audio-sgtl5000"; + nvidia,model = "Toradex Apalis T30 SGTL5000"; + nvidia,audio-routing = + "Headphone Jack", "HP_OUT", + "LINE_IN", "Line In Jack", + "MIC_IN", "Mic Jack"; + nvidia,i2s-controller = <&tegra_i2s2>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA30_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt deleted file mode 100644 index ef1fe7358279..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt +++ /dev/null @@ -1,21 +0,0 @@ -NVIDIA Tegra audio complex for TrimSlice - -Required properties: -- compatible : "nvidia,tegra-audio-trimslice" -- clocks : Must contain an entry for each entry in clock-names. -- clock-names : Must include the following entries: - "pll_a" (The Tegra clock of that name), - "pll_a_out0" (The Tegra clock of that name), - "mclk" (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-trimslice"; - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&codec>; - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml new file mode 100644 index 000000000000..8c87cd166238 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-trimslice.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with TrimSlice CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + const: nvidia,tegra-audio-trimslice + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-trimslice"; + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt deleted file mode 100644 index 96f6a57dd6b4..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt +++ /dev/null @@ -1,40 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8753" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the WM8753's pins as documented in the binding for the WM8753, - and the jacks on the board: - - * Headphone Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8753 audio codec -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8753-whistler", - "nvidia,tegra-audio-wm8753" - nvidia,model = "tegra-wm8753-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "LOUT1", - "Headphone Jack", "ROUT1"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8753>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml new file mode 100644 index 000000000000..a5b431d7d0c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm8753.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM8753 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8753(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm8753 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Mic Jack" + + # CODEC Pins + - LOUT1 + - LOUT2 + - ROUT1 + - ROUT2 + - MONO1 + - MONO2 + - OUT3 + - OUT4 + - LINE1 + - LINE2 + - RXP + - RXN + - ACIN + - ACOP + - MIC1N + - MIC1 + - MIC2N + - MIC2 + - "Mic Bias" + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8753-whistler", + "nvidia,tegra-audio-wm8753"; + nvidia,model = "tegra-wm8753-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "LOUT1", + "Headphone Jack", "ROUT1"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8753>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt deleted file mode 100644 index bbd581a8c5bc..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ /dev/null @@ -1,62 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8903" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins (documented in the WM8903 binding document), - and the jacks on the board: - - * Headphone Jack - * Int Spk - * Mic Jack - * Int Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Optional properties: -- nvidia,spkr-en-gpios : The GPIO that enables the speakers -- nvidia,hp-mute-gpios : The GPIO that mutes the headphones -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone -- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone -- nvidia,headset : The Mic Jack represents state of the headset microphone pin - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8903-harmony", - "nvidia,tegra-audio-wm8903" - nvidia,model = "tegra-wm8903-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "HPOUTR", - "Headphone Jack", "HPOUTL", - "Int Spk", "ROP", - "Int Spk", "RON", - "Int Spk", "LOP", - "Int Spk", "LON", - "Mic Jack", "MICBIAS", - "IN1L", "Mic Jack"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8903>; - - nvidia,spkr-en-gpios = <&codec 2 0>; - nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */ - nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */ - nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */ - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml new file mode 100644 index 000000000000..1b836acab980 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm8903.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM8903 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8903(-[a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm8903 + - items: + - pattern: ad,tegra-audio-plutux + - const: nvidia,tegra-audio-wm8903 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone Jack" + - "Int Spk" + - "Mic Jack" + - "Int Mic" + + # CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - DMICDAT + - HPOUTL + - HPOUTR + - LINEOUTL + - LINEOUTR + - LOP + - LON + - ROP + - RON + - MICBIAS + +required: + - nvidia,i2s-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm8903-harmony", + "nvidia,tegra-audio-wm8903"; + nvidia,model = "tegra-wm8903-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "HPOUTR", + "Headphone Jack", "HPOUTL", + "Int Spk", "ROP", + "Int Spk", "RON", + "Int Spk", "LOP", + "Int Spk", "LON", + "Mic Jack", "MICBIAS", + "IN1L", "Mic Jack"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8903>; + + nvidia,spkr-en-gpios = <&codec 2 0>; + nvidia,hp-det-gpios = <&gpio 178 0>; + nvidia,int-mic-en-gpios = <&gpio 184 0>; + nvidia,ext-mic-en-gpios = <&gpio 185 0>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt deleted file mode 100644 index 436f6cd9d07c..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt +++ /dev/null @@ -1,60 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm9712" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the WM9712's pins, and the jacks on the board: - - WM9712 pins: - - * MONOOUT - * HPOUTL - * HPOUTR - * LOUT2 - * ROUT2 - * OUT3 - * LINEINL - * LINEINR - * PHONE - * PCBEEP - * MIC1 - * MIC2 - * Mic Bias - - Board connectors: - - * Headphone - * LineIn - * Mic - -- nvidia,ac97-controller : The phandle of the Tegra AC97 controller - - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm9712-colibri_t20", - "nvidia,tegra-audio-wm9712"; - nvidia,model = "Toradex Colibri T20"; - - nvidia,audio-routing = - "Headphone", "HPOUTL", - "Headphone", "HPOUTR", - "LineIn", "LINEINL", - "LineIn", "LINEINR", - "Mic", "MIC1"; - - nvidia,ac97-controller = <&ac97>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml new file mode 100644 index 000000000000..a1448283344b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-wm9712.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra audio complex with WM9712 CODEC + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +allOf: + - $ref: nvidia,tegra-audio-common.yaml# + +properties: + compatible: + items: + - pattern: '^[a-z0-9]+,tegra-audio-wm9712([-_][a-z0-9]+)+$' + - const: nvidia,tegra-audio-wm9712 + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board. + minItems: 2 + items: + enum: + # Board Connectors + - "Headphone" + - "LineIn" + - "Mic" + + # CODEC Pins + - MONOOUT + - HPOUTL + - HPOUTR + - LOUT2 + - ROUT2 + - OUT3 + - LINEINL + - LINEINR + - PHONE + - PCBEEP + - MIC1 + - MIC2 + - "Mic Bias" + +required: + - nvidia,ac97-controller + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "nvidia,tegra-audio-wm9712-colibri_t20", + "nvidia,tegra-audio-wm9712"; + nvidia,model = "Toradex Colibri T20"; + + nvidia,audio-routing = + "Headphone", "HPOUTL", + "Headphone", "HPOUTR", + "LineIn", "LINEINL", + "LineIn", "LINEINR", + "Mic", "MIC1"; + + nvidia,ac97-controller = <&ac97>; + + clocks = <&clk 112>, <&clk 113>, <&clk 93>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; -- 2.33.0

5 6

[PATCH 0/8] Changes to SOF kcontrol data set/get ops
by Ranjani Sridharan 16 Dec '21

16 Dec '21

This set of patches deals with modifications to the signature of kcontrol get/set data functions to make them more intuitive. The last patch deals with initializing the binary control data size after boot up. Peter Ujfalusi (7): ASoC: SOF: ipc: Rename send parameter in snd_sof_ipc_set_get_comp_data() ASoC: SOF: Drop ipc_cmd parameter for snd_sof_ipc_set_get_comp_data() ASoC: SOF: topology: Set control_data->cmd alongside scontrol->cmd ASoC: SOF: Drop ctrl_cmd parameter for snd_sof_ipc_set_get_comp_data() ASoC: SOF: sof-audio: Drop the `cmd` member from struct snd_sof_control ASoC: SOF: control: Do not handle control notification with component type ASoC: SOF: Drop ctrl_type parameter for snd_sof_ipc_set_get_comp_data() Ranjani Sridharan (1): ASoC: SOF: topology: read back control data from DSP sound/soc/sof/control.c | 61 +++++++++++---------------------------- sound/soc/sof/ipc.c | 49 +++++++++++++++---------------- sound/soc/sof/sof-audio.c | 33 ++++++++++----------- sound/soc/sof/sof-audio.h | 7 +---- sound/soc/sof/topology.c | 10 +++---- 5 files changed, 62 insertions(+), 98 deletions(-) -- 2.25.1

2 9

[PATCH] ASoC: AMD: fix depend/select mistake on SND_AMD_ACP_CONFIG
by Pierre-Louis Bossart 16 Dec '21

16 Dec '21

on i386 or x86_64: when # CONFIG_ACPI is not set, so SND_SOC_ACPI is not set: WARNING: unmet direct dependencies detected for SND_AMD_ACP_CONFIG Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_ACPI [=n] Selected by [y]: - SND_SOC_AMD_ACP_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && X86 [=y] && PCI [=y] This problem is due to the unconditional selection of SND_AMD_ACP_CONFIG in other options. Using 'depends on' solved an initial problem but exposed another, let's use select instead. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Fixes: d9b994cd7641 ('ASoC: AMD: acp-config: fix missing dependency on SND_SOC_ACPI') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Randy Dunlap <rdunlap(a)infradead.org> # build-tested Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)intel.com> Reviewed-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index bcfeb3fc2592..7a9e45094f37 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -98,7 +98,7 @@ config SND_SOC_AMD_YC_MACH config SND_AMD_ACP_CONFIG tristate "AMD ACP configuration selection" - depends on SND_SOC_ACPI + select SND_SOC_ACPI if ACPI help This option adds an auto detection to determine which ACP driver modules to use -- 2.25.1

2 1

[PATCH v2] ASoC: dt-bindings: audio-graph-port: enable both flag/phandle for bitclock/frame-master
by Kuninori Morimoto 16 Dec '21

16 Dec '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_daifmt_parse_clock_provider_raw() is handling both bitclock/frame-master, and is supporting both flag/phandle. Current DT is assuming it is flag style. This patch allows both case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/audio-graph-port.yaml | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 43e7f86e3b23..7d0248be08c5 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -42,10 +42,15 @@ patternProperties: $ref: /schemas/types.yaml#/definitions/flag frame-master: description: Indicates dai-link frame master. - $ref: /schemas/types.yaml#/definitions/phandle + anyOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle bitclock-master: description: Indicates dai-link bit clock master - $ref: /schemas/types.yaml#/definitions/phandle + anyOf: + - $ref: /schemas/types.yaml#/definitions/flag + - $ref: /schemas/types.yaml#/definitions/phandle + dai-format: description: audio format. items: -- 2.25.1

2 2

[PATCH 1/2] ASoC: dt-bindings: codecs: Add bindings for ak4375
by Vincent Knecht 15 Dec '21

15 Dec '21

AK4375 is an audio DAC with headphones amplifier controlled via I2C. Add simple device tree bindings that describe how to set it up. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- .../devicetree/bindings/sound/ak4375.yaml | 56 +++++++++++++++++++ 1 file changed, 56 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4375.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4375.yaml b/Documentation/devicetree/bindings/sound/ak4375.yaml new file mode 100644 index 000000000000..295972801872 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4375.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4375.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4375 DAC and headphones amplifier Device Tree Bindings + +maintainers: + - Vincent Knecht <vincent.knecht(a)mailoo.org> + +properties: + compatible: + const: asahi-kasei,ak4375 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + avdd-supply: + description: optional regulator phandle for the AVDD power supply. + + tvdd-supply: + description: regulator phandle for the TVDD power supply. + + pdn-gpios: + description: optional GPIO to set the PDN pin. + +required: + - compatible + - reg + - '#sound-dai-cells' + - tvdd-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + headphones: audio-codec@10 { + compatible = "asahi-kasei,ak4375"; + reg = <0x10>; + avdd-supply = <&reg_headphones_avdd>; + tvdd-supply = <&pm8916_l6>; + pdn-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&headphones_pdn_default>; + #sound-dai-cells = <0>; + }; + }; -- 2.31.1

3 6

[PATCH] ASoC: rt5663: Handle device_property_read_u32_array error codes
by Jiasheng Jiang 15 Dec '21

15 Dec '21

The return value of device_property_read_u32_array() is not always 0. To catch the exception in case that devm_kzalloc failed and the rt5663->imp_table was NULL, which caused the failure of device_property_read_u32_array. Fixes: 450f0f6a8fb4 ("ASoC: rt5663: Add the manual offset field to compensate the DC offset") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/codecs/rt5663.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index be9fc58ff681..ee09ccd448dc 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3461,6 +3461,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) { int table_size; + int ret; device_property_read_u32(dev, "realtek,dc_offset_l_manual", &rt5663->pdata.dc_offset_l_manual); @@ -3477,9 +3478,11 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) table_size = sizeof(struct impedance_mapping_table) * rt5663->pdata.impedance_sensing_num; rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); - device_property_read_u32_array(dev, + ret = device_property_read_u32_array(dev, "realtek,impedance_sensing_table", (u32 *)rt5663->imp_table, table_size); + if (ret) + return ret; } return 0; @@ -3504,8 +3507,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, if (pdata) rt5663->pdata = *pdata; - else - rt5663_parse_dp(rt5663, &i2c->dev); + else { + ret = rt5663_parse_dp(rt5663, &i2c->dev); + if (ret) + return ret; + } for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) rt5663->supplies[i].supply = rt5663_supply_names[i]; -- 2.25.1

2 1

[PATCH] ASoC: SOF: OF: Avoid reverse module dependency
by Daniel Baluta 15 Dec '21

15 Dec '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> Similar with commit 8a49cd11e68ed0 ("ASoC: SOF: ACPI: avoid reverse module dependency") we will be having hardware specific drivers that link against a common "helper" framework. sof-of-dev.c becomes a library with the interface defined in the newly created file sof-of-dev.h. This is the final step started with Kconfig simplification in commit 7548a391c53ca ("ASoC: SOF: i.MX: simplify Kconfig") Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx-ops.h | 10 ---- sound/soc/sof/imx/imx8.c | 43 ++++++++++++-- sound/soc/sof/imx/imx8m.c | 31 ++++++++++- sound/soc/sof/mediatek/mediatek-ops.h | 8 --- sound/soc/sof/mediatek/mt8195/mt8195.c | 31 ++++++++++- sound/soc/sof/sof-of-dev.c | 77 +++----------------------- sound/soc/sof/sof-of-dev.h | 17 ++++++ 7 files changed, 118 insertions(+), 99 deletions(-) delete mode 100644 sound/soc/sof/imx/imx-ops.h delete mode 100644 sound/soc/sof/mediatek/mediatek-ops.h create mode 100644 sound/soc/sof/sof-of-dev.h diff --git a/sound/soc/sof/imx/imx-ops.h b/sound/soc/sof/imx/imx-ops.h deleted file mode 100644 index 24235ef8c8fa..000000000000 --- a/sound/soc/sof/imx/imx-ops.h +++ /dev/null @@ -1,10 +0,0 @@ -/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ - -#ifndef __IMX_OPS_H__ -#define __IMX_OPS_H__ - -extern struct snd_sof_dsp_ops sof_imx8_ops; -extern struct snd_sof_dsp_ops sof_imx8x_ops; -extern struct snd_sof_dsp_ops sof_imx8m_ops; - -#endif diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 00b2bb5fd6ae..099b4356122c 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -21,8 +21,8 @@ #include <linux/firmware/imx/svc/misc.h> #include <dt-bindings/firmware/imx/rsrc.h> #include "../ops.h" +#include "../sof-of-dev.h" #include "imx-common.h" -#include "imx-ops.h" /* DSP memories */ #define IRAM_OFFSET 0x10000 @@ -487,7 +487,7 @@ static int imx8_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8_ops = { +static const struct snd_sof_dsp_ops sof_imx8_ops = { /* probe and remove */ .probe = imx8_probe, .remove = imx8_remove, @@ -548,10 +548,9 @@ struct snd_sof_dsp_ops sof_imx8_ops = { .set_power_state = imx8_dsp_set_power_state, }; -EXPORT_SYMBOL(sof_imx8_ops); /* i.MX8X ops */ -struct snd_sof_dsp_ops sof_imx8x_ops = { +static const struct snd_sof_dsp_ops sof_imx8x_ops = { /* probe and remove */ .probe = imx8_probe, .remove = imx8_remove, @@ -612,7 +611,41 @@ struct snd_sof_dsp_ops sof_imx8x_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP }; -EXPORT_SYMBOL(sof_imx8x_ops); + +static struct sof_dev_desc sof_of_imx8qxp_desc = { + .default_fw_path = "imx/sof", + .default_tplg_path = "imx/sof-tplg", + .default_fw_filename = "sof-imx8x.ri", + .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", + .ops = &sof_imx8x_ops, +}; + +static struct sof_dev_desc sof_of_imx8qm_desc = { + .default_fw_path = "imx/sof", + .default_tplg_path = "imx/sof-tplg", + .default_fw_filename = "sof-imx8.ri", + .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", + .ops = &sof_imx8_ops, +}; + +static const struct of_device_id sof_of_imx8_ids[] = { + { .compatible = "fsl,imx8qxp-dsp", .data = &sof_of_imx8qxp_desc}, + { .compatible = "fsl,imx8qm-dsp", .data = &sof_of_imx8qm_desc}, + { } +}; +MODULE_DEVICE_TABLE(of, sof_of_imx8_ids); + +/* DT driver definition */ +static struct platform_driver snd_sof_of_imx8_driver = { + .probe = sof_of_probe, + .remove = sof_of_remove, + .driver = { + .name = "sof-audio-of-imx8", + .pm = &sof_of_pm, + .of_match_table = sof_of_imx8_ids, + }, +}; +module_platform_driver(snd_sof_of_imx8_driver); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index b7df655c49c9..c026caea4c8b 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -20,8 +20,8 @@ #include <linux/firmware/imx/dsp.h> #include "../ops.h" +#include "../sof-of-dev.h" #include "imx-common.h" -#include "imx-ops.h" #define MBOX_OFFSET 0x800000 #define MBOX_SIZE 0x1000 @@ -411,7 +411,7 @@ static int imx8m_dsp_suspend(struct snd_sof_dev *sdev, unsigned int target_state } /* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8m_ops = { +static const struct snd_sof_dsp_ops sof_imx8m_ops = { /* probe and remove */ .probe = imx8m_probe, .remove = imx8m_remove, @@ -470,7 +470,32 @@ struct snd_sof_dsp_ops sof_imx8m_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, }; -EXPORT_SYMBOL(sof_imx8m_ops); + +static struct sof_dev_desc sof_of_imx8mp_desc = { + .default_fw_path = "imx/sof", + .default_tplg_path = "imx/sof-tplg", + .default_fw_filename = "sof-imx8m.ri", + .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", + .ops = &sof_imx8m_ops, +}; + +static const struct of_device_id sof_of_imx8m_ids[] = { + { .compatible = "fsl,imx8mp-dsp", .data = &sof_of_imx8mp_desc}, + { } +}; +MODULE_DEVICE_TABLE(of, sof_of_imx8m_ids); + +/* DT driver definition */ +static struct platform_driver snd_sof_of_imx8m_driver = { + .probe = sof_of_probe, + .remove = sof_of_remove, + .driver = { + .name = "sof-audio-of-imx8m", + .pm = &sof_of_pm, + .of_match_table = sof_of_imx8m_ids, + }, +}; +module_platform_driver(snd_sof_of_imx8m_driver); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/mediatek/mediatek-ops.h b/sound/soc/sof/mediatek/mediatek-ops.h deleted file mode 100644 index e0ffa69ecb0c..000000000000 --- a/sound/soc/sof/mediatek/mediatek-ops.h +++ /dev/null @@ -1,8 +0,0 @@ -/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ - -#ifndef __MEDIATEK_OPS_H__ -#define __MEDIATEK_OPS_H__ - -extern const struct snd_sof_dsp_ops sof_mt8195_ops; - -#endif diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index c719ba470620..3ab12f352935 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -21,9 +21,9 @@ #include <sound/sof.h> #include <sound/sof/xtensa.h> #include "../../ops.h" +#include "../../sof-of-dev.h" #include "../../sof-audio.h" #include "../adsp_helper.h" -#include "../mediatek-ops.h" #include "mt8195.h" #include "mt8195-clk.h" @@ -388,7 +388,7 @@ static struct snd_soc_dai_driver mt8195_dai[] = { }; /* mt8195 ops */ -const struct snd_sof_dsp_ops sof_mt8195_ops = { +static const struct snd_sof_dsp_ops sof_mt8195_ops = { /* probe and remove */ .probe = mt8195_dsp_probe, .remove = mt8195_dsp_remove, @@ -432,7 +432,32 @@ const struct snd_sof_dsp_ops sof_mt8195_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, }; -EXPORT_SYMBOL(sof_mt8195_ops); + +static const struct sof_dev_desc sof_of_mt8195_desc = { + .default_fw_path = "mediatek/sof", + .default_tplg_path = "mediatek/sof-tplg", + .default_fw_filename = "sof-mt8195.ri", + .nocodec_tplg_filename = "sof-mt8195-nocodec.tplg", + .ops = &sof_mt8195_ops, +}; + +static const struct of_device_id sof_of_mt8195_ids[] = { + { .compatible = "mediatek,mt8195-dsp", .data = &sof_of_mt8195_desc}, + { } +}; +MODULE_DEVICE_TABLE(of, sof_of_mt8195_ids); + +/* DT driver definition */ +static struct platform_driver snd_sof_of_mt8195_driver = { + .probe = sof_of_probe, + .remove = sof_of_remove, + .driver = { + .name = "sof-audio-of-mt8195", + .pm = &sof_of_pm, + .of_match_table = sof_of_mt8195_ids, + }, +}; +module_platform_driver(snd_sof_of_mt8195_driver); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index b0089698eecb..e3718638f9ce 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -11,9 +11,8 @@ #include <linux/pm_runtime.h> #include <sound/sof.h> +#include "sof-of-dev.h" #include "ops.h" -#include "imx/imx-ops.h" -#include "mediatek/mediatek-ops.h" static char *fw_path; module_param(fw_path, charp, 0444); @@ -23,51 +22,14 @@ static char *tplg_path; module_param(tplg_path, charp, 0444); MODULE_PARM_DESC(tplg_path, "alternate path for SOF topology."); -/* platform specific devices */ -#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) -static struct sof_dev_desc sof_of_imx8qxp_desc = { - .default_fw_path = "imx/sof", - .default_tplg_path = "imx/sof-tplg", - .default_fw_filename = "sof-imx8x.ri", - .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", - .ops = &sof_imx8x_ops, -}; - -static struct sof_dev_desc sof_of_imx8qm_desc = { - .default_fw_path = "imx/sof", - .default_tplg_path = "imx/sof-tplg", - .default_fw_filename = "sof-imx8.ri", - .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", - .ops = &sof_imx8_ops, -}; -#endif - -#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8M) -static struct sof_dev_desc sof_of_imx8mp_desc = { - .default_fw_path = "imx/sof", - .default_tplg_path = "imx/sof-tplg", - .default_fw_filename = "sof-imx8m.ri", - .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", - .ops = &sof_imx8m_ops, -}; -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_MT8195) -static const struct sof_dev_desc sof_of_mt8195_desc = { - .default_fw_path = "mediatek/sof", - .default_tplg_path = "mediatek/sof-tplg", - .default_fw_filename = "sof-mt8195.ri", - .nocodec_tplg_filename = "sof-mt8195-nocodec.tplg", - .ops = &sof_mt8195_ops, -}; -#endif - -static const struct dev_pm_ops sof_of_pm = { +const struct dev_pm_ops sof_of_pm = { .prepare = snd_sof_prepare, .complete = snd_sof_complete, SET_SYSTEM_SLEEP_PM_OPS(snd_sof_suspend, snd_sof_resume) SET_RUNTIME_PM_OPS(snd_sof_runtime_suspend, snd_sof_runtime_resume, NULL) }; +EXPORT_SYMBOL(sof_of_pm); static void sof_of_probe_complete(struct device *dev) { @@ -79,7 +41,7 @@ static void sof_of_probe_complete(struct device *dev) pm_runtime_enable(dev); } -static int sof_of_probe(struct platform_device *pdev) +int sof_of_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; const struct sof_dev_desc *desc; @@ -120,8 +82,9 @@ static int sof_of_probe(struct platform_device *pdev) /* call sof helper for DSP hardware probe */ return snd_sof_device_probe(dev, sof_pdata); } +EXPORT_SYMBOL(sof_of_probe); -static int sof_of_remove(struct platform_device *pdev) +int sof_of_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); @@ -130,32 +93,6 @@ static int sof_of_remove(struct platform_device *pdev) return 0; } - -static const struct of_device_id sof_of_ids[] = { -#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) - { .compatible = "fsl,imx8qxp-dsp", .data = &sof_of_imx8qxp_desc}, - { .compatible = "fsl,imx8qm-dsp", .data = &sof_of_imx8qm_desc}, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8M) - { .compatible = "fsl,imx8mp-dsp", .data = &sof_of_imx8mp_desc}, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_MT8195) - { .compatible = "mediatek,mt8195-dsp", .data = &sof_of_mt8195_desc}, -#endif - { } -}; -MODULE_DEVICE_TABLE(of, sof_of_ids); - -/* DT driver definition */ -static struct platform_driver snd_sof_of_driver = { - .probe = sof_of_probe, - .remove = sof_of_remove, - .driver = { - .name = "sof-audio-of", - .pm = &sof_of_pm, - .of_match_table = sof_of_ids, - }, -}; -module_platform_driver(snd_sof_of_driver); +EXPORT_SYMBOL(sof_of_remove); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/sof-of-dev.h b/sound/soc/sof/sof-of-dev.h new file mode 100644 index 000000000000..4e0f6588dad9 --- /dev/null +++ b/sound/soc/sof/sof-of-dev.h @@ -0,0 +1,17 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright 2021 NXP + */ + +#ifndef __SOUND_SOC_SOF_OF_H +#define __SOUND_SOC_SOF_OF_H + +extern const struct dev_pm_ops sof_of_pm; + +int sof_of_probe(struct platform_device *pdev); +int sof_of_remove(struct platform_device *pdev); + +#endif -- 2.27.0

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Re: [PATCH] ASoC: tas2770: Fix setting of high sample rates
by Mark Brown 15 Dec '21

15 Dec '21

On Mon, 06 Dec 2021 22:45:43 +0000, Martin Povišer wrote: > Although the codec advertises support for 176.4 and 192 ksps, without > this fix setting those sample rates fails with EINVAL at hw_params time. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-linus Thanks! [1/1] ASoC: tas2770: Fix setting of high sample rates commit: 80d5be1a057e05f01d66e986cfd34d71845e5190 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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[PATCH v2 0/2] ASoC: meson: aiu: two fixes (for 5.16)
by Martin Blumenstingl 15 Dec '21

15 Dec '21

In this series I am proposing two fixes for the "aiu" driver, used on Amlogic Meson8, Meson8b, Meson8m2, GXBB, GXL and GXM SoCs. The first patch is the result of me trying to understand the way how we get the DMA buffer and address for the audio data. I'm not an expert in terms of DMA. From what I understand we need to inform DMA core about the limitations of the hardware. In case of AIU it's DMA address registers only take 32 bits, so DMA core should be aware of this. The second patch is what I could come up with to fix the infamous I2S buffer underrun issue, also called the "machine gun noise" (MGN) bug. After a lot of testing, debugging and comparing vendor code with the upstream "aiu" driver I have come up with this fix. I have written down my thoughts in the description of that patch. To be clear: these are my thoughts, unfortunately I have no way of proving this other than asking other people to test this patch (off-list I have already received positive feedback along with confirmation that both 2-ch and 6-ch audio are still working fine. Even with Kodi's menu - which is an easy way to reproduce the MGN bug - sound output is fine with this patch). @Mark, @Liam: It would be great if at least the second patch could be queued as a fix for "for-5.16" as this solves a long standing buffer underrun. This issue is nasty because it can occur at any time and it can even loop forever. Christian provided an example with the speakers on low volume: [1]. Imagine the same sound on "medium" volume after playing a movie. Changes since v1 at [0]: - Dropped a paragraph about FIFO IRQs from the second patch because Jerome has a valid point that this behavior can either mean that AIU_I2S_MISC is related to the FIFO or the FIFO consumer (= encoder) - Add a bit of documentation explaining why we set the AIU_I2S_MISC_FORCE_LEFT_RIGHT bit (affects patch #2) - Collect Christian's and Geraldo's Tested-by as well as Jerome's Acked-by for patch #2 (thanks to all three of you!) - Cc linux-stable for the second patch - Dropped RFC prefix [0] https://chewitt.libreelec.tv/testing/wp2_audio_noise.mov [1] https://patchwork.kernel.org/project/linux-amlogic/cover/20211205180816.208… Martin Blumenstingl (2): ASoC: meson: aiu: fifo: Add missing dma_coerce_mask_and_coherent() ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s sound/soc/meson/aiu-encoder-i2s.c | 33 ------------------------------- sound/soc/meson/aiu-fifo-i2s.c | 19 ++++++++++++++++++ sound/soc/meson/aiu-fifo.c | 6 ++++++ 3 files changed, 25 insertions(+), 33 deletions(-) -- 2.34.1

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[PATCH v7 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 15 Dec '21

15 Dec '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v6->v7: - Minor review fixes in devicetree binding file and related include file. Changes v5->v6: - Added missing "depends on I2C" in the Kconfig (reported by lkp(a)intel.com). - Fold include file into source as it is not needed elsewhere. - Fix SPDX-License-Identifier which was using a deprecated license. - Clean up authorship information in file header, removing history. - Remove useless GPIOLIB ifdefs. - Use devm_gpiod_get for reset gpio rather than devm_gpio_request etc. - Handle EPROBE_DEFER properly for reset gpio request. - Use dev_err_probe rather than explicit EPROBE_DEFER checking. - Properly check value from DT parsing for mic bias and codec gpio pins. No changes to the devicetree bindings. Changes v4->v5: Yet another fix based on kernel test robot (lkp(a)intel.com), this time a legitimate issue: the return value from of_get_named_gpio_flags() was crammed into an unsigned int so any potential negative error codes were lost. No changes to the devicetree bindings. Changes v3->v4: More fixes reported by kernel test robot (lkp(a)intel.com). Don't know why these weren't reported the first time. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1311 +++++++++++++++++ 5 files changed, 1486 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH v4 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 15 Dec '21

15 Dec '21

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 27 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 525 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 68 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 147 ++++ sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 353 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2049 insertions(+), 1092 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

4 13

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for a HP ProBook
by Jeremy Szu 15 Dec '21

15 Dec '21

There is a HP ProBook which using ALC236 codec and need the ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make mute LED and micmute LED work. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- 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 3599f4c85ebf..e97b3cd1577b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8705,6 +8705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.33.0

2 1

Cant't open mixer and no sound , helem (MIXER, 'Front Mic Boots Valume', 0, 1, 0) appears twice or more
by GitHub issues - edited 15 Dec '21

15 Dec '21

alsa-project/alsa-lib issue #205 was edited from WolsYang: **I had try the way in this issue, but not work for me : (** **helem (MIXER,'Master Playback Volume',0,1,0) appears twice or more cannot load mixer controls: Invalid argument**#47 **In default I only can play sound with Back port** **Here is my computer audio port look like.** |codec | ALC888-VD | ALC888-VD | |:-------------|:----------------:|:----------------:| |position | Back | Front | |Function | 0:Line Out, 1:MIC | 0:Line Out, 1:MIC| |Port Num | 2 | 2 | |HW:dev|0:0|0:2| |/sys/class/cound|hwC0D0|hwC0D1| **OS:Ubuntu 20.04.2LTS** **$ uname -r** ``` 5.13.0-051300-generic ``` **$ amixer** ``` ALSA lib simple_none.c:1544:(simple_add1) helem (MIXER,'Front Mic Boots Valume',0,1,0) appears twice or more amixer: Mixer default load error: Invalid aggument ``` **$ aplay -l** ``` **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` **$ amixer -c 0 controls** ``` numid=18,iface=CARD,name='Front Line Out Front Phantom Jack' numid=16,iface=CARD,name='Front Mic Jack' numid=76,iface=CARD,name='Front Mic Jack',index=1 numid=41,iface=CARD,name='HDMI/DP,pcm=3 Jack' numid=47,iface=CARD,name='HDMI/DP,pcm=7 Jack' numid=53,iface=CARD,name='HDMI/DP,pcm=8 Jack' numid=78,iface=CARD,name='Headphone Front Jack' numid=79,iface=CARD,name='Headphone Surround Jack' numid=19,iface=CARD,name='Line Out Surround Phantom Jack' numid=17,iface=CARD,name='Rear Mic Jack' numid=77,iface=CARD,name='Rear Mic Jack',index=1 numid=15,iface=MIXER,name='Master Playback Switch' numid=14,iface=MIXER,name='Master Playback Volume' numid=64,iface=MIXER,name='Headphone Playback Switch' numid=66,iface=MIXER,name='Headphone Playback Switch',index=1 numid=63,iface=MIXER,name='Headphone Playback Volume' numid=65,iface=MIXER,name='Headphone Playback Volume',index=1 numid=62,iface=MIXER,name='PCM Playback Volume' numid=12,iface=MIXER,name='Front Mic Boost Volume' numid=7,iface=MIXER,name='Front Mic Playback Switch' numid=6,iface=MIXER,name='Front Mic Playback Volume' numid=74,iface=MIXER,name='Front Mic Boost Volume',device=1 numid=69,iface=MIXER,name='Front Mic Playback Switch',device=1 numid=68,iface=MIXER,name='Front Mic Playback Volume',device=1 numid=2,iface=MIXER,name='Front Playback Switch' numid=1,iface=MIXER,name='Front Playback Volume' numid=4,iface=MIXER,name='Surround Playback Switch' numid=3,iface=MIXER,name='Surround Playback Volume' numid=11,iface=MIXER,name='Capture Switch' numid=73,iface=MIXER,name='Capture Switch',device=1 numid=10,iface=MIXER,name='Capture Volume' numid=72,iface=MIXER,name='Capture Volume',device=1 numid=5,iface=MIXER,name='Loopback Mixing' numid=67,iface=MIXER,name='Loopback Mixing',device=1 numid=42,iface=MIXER,name='IEC958 Playback Con Mask' numid=48,iface=MIXER,name='IEC958 Playback Con Mask',index=1 numid=54,iface=MIXER,name='IEC958 Playback Con Mask',index=2 numid=43,iface=MIXER,name='IEC958 Playback Pro Mask' numid=49,iface=MIXER,name='IEC958 Playback Pro Mask',index=1 numid=55,iface=MIXER,name='IEC958 Playback Pro Mask',index=2 numid=44,iface=MIXER,name='IEC958 Playback Default' numid=50,iface=MIXER,name='IEC958 Playback Default',index=1 numid=56,iface=MIXER,name='IEC958 Playback Default',index=2 numid=45,iface=MIXER,name='IEC958 Playback Switch' numid=51,iface=MIXER,name='IEC958 Playback Switch',index=1 numid=57,iface=MIXER,name='IEC958 Playback Switch',index=2 numid=13,iface=MIXER,name='Rear Mic Boost Volume' numid=9,iface=MIXER,name='Rear Mic Playback Switch' numid=8,iface=MIXER,name='Rear Mic Playback Volume' numid=75,iface=MIXER,name='Rear Mic Boost Volume',device=1 numid=71,iface=MIXER,name='Rear Mic Playback Switch',device=1 numid=70,iface=MIXER,name='Rear Mic Playback Volume',device=1 numid=21,iface=PCM,name='Capture Channel Map' numid=20,iface=PCM,name='Playback Channel Map' numid=81,iface=PCM,name='Capture Channel Map',device=2 numid=80,iface=PCM,name='Playback Channel Map',device=2 numid=46,iface=PCM,name='ELD',device=3 numid=59,iface=PCM,name='Playback Channel Map',device=3 numid=52,iface=PCM,name='ELD',device=7 numid=60,iface=PCM,name='Playback Channel Map',device=7 numid=58,iface=PCM,name='ELD',device=8 numid=61,iface=PCM,name='Playback Channel Map',device=8 ``` **$ pactl list modules** ``` Card #0 Name: alsa_card.pci-0000_00_1b.0 Driver: module-alsa-card.c Owner Module: 7 Properties: alsa.card = "0" alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0x81410000 irq 132" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.id = "2284" device.product.name = "Atom/Celeron/Pentium Processor x5-E8000/J3xxx/N3xxx Series High Definition Audio Controller" device.form_factor = "internal" device.string = "0" device.description = "Built-in Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) output:analog-surround-40: Analog Surround 4.0 Output (sinks: 1, sources: 0, priority: 1200, available: yes) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 1265, available: yes) output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 5900, available: yes) output:hdmi-stereo+input:analog-stereo: Digital Stereo (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5965, available: yes) output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround+input:analog-stereo: Digital Surround 5.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-surround71: Digital Surround 7.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround71+input:analog-stereo: Digital Surround 7.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-stereo-extra1: Digital Stereo (HDMI 2) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra1+input:analog-stereo: Digital Stereo (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) output:hdmi-surround-extra1: Digital Surround 5.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround-extra1+input:analog-stereo: Digital Surround 5.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-surround71-extra1: Digital Surround 7.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround71-extra1+input:analog-stereo: Digital Surround 7.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-stereo-extra2: Digital Stereo (HDMI 3) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra2+input:analog-stereo: Digital Stereo (HDMI 3) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/205 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Cant't open mixer and no sound , helem (MIXER, 'Front Mic Boots Valume', 0, 1, 0) appears twice or more
by GitHub issues - opened 15 Dec '21

15 Dec '21

alsa-project/alsa-lib issue #205 was opened from WolsYang: **I had try the way in this issue, but not work for me : (** **helem (MIXER,'Master Playback Volume',0,1,0) appears twice or more cannot load mixer controls: Invalid argument**#47 **In default I only can play sound with Back port** **Here is my computer audio port look like.** |codec | ALC888-VD | ALC888-VD | |:-------------|:----------------:|:----------------:| |position | Back | Front | |Function | 0:Line Out, 1:MIC | 0:Line Out, 1:MIC| |Port Num | 2 | 2 | |HW:dev|0:0|0:2| |/sys/class/cound|hwC0D0|hwC0D1| **OS:Ubuntu 20.04.2LTS** **$ uname -r** ``` 5.13.0-051300-generic ``` **$ amixer** ``` ALSA lib simple_none.c:1544:(simple_add1) helem (MIXER,'Front Mic Boots Valume',0,1,0) appears twice or more amixer: Mixer default load error: Invalid aggument ``` **$ aplay -l** ``` **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` **$ pactl list modules** ``` Card #0 Name: alsa_card.pci-0000_00_1b.0 Driver: module-alsa-card.c Owner Module: 7 Properties: alsa.card = "0" alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0x81410000 irq 132" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.id = "2284" device.product.name = "Atom/Celeron/Pentium Processor x5-E8000/J3xxx/N3xxx Series High Definition Audio Controller" device.form_factor = "internal" device.string = "0" device.description = "Built-in Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) output:analog-surround-40: Analog Surround 4.0 Output (sinks: 1, sources: 0, priority: 1200, available: yes) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 1265, available: yes) output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 5900, available: yes) output:hdmi-stereo+input:analog-stereo: Digital Stereo (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5965, available: yes) output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround+input:analog-stereo: Digital Surround 5.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-surround71: Digital Surround 7.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround71+input:analog-stereo: Digital Surround 7.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-stereo-extra1: Digital Stereo (HDMI 2) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra1+input:analog-stereo: Digital Stereo (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) output:hdmi-surround-extra1: Digital Surround 5.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround-extra1+input:analog-stereo: Digital Surround 5.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-surround71-extra1: Digital Surround 7.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround71-extra1+input:analog-stereo: Digital Surround 7.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-stereo-extra2: Digital Stereo (HDMI 3) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra2+input:analog-stereo: Digital Stereo (HDMI 3) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/205 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: qdsp6: Fix an IS_ERR() vs NULL bug
by Miaoqian Lin 15 Dec '21

15 Dec '21

The function gpr_alloc_port return ERR_PTR on errors, it doesn't return null. Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/qcom/qdsp6/q6apm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 13598ef5bacb..3e007d609a9b 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -630,9 +630,9 @@ struct q6apm_graph *q6apm_graph_open(struct device *dev, q6apm_cb cb, init_waitqueue_head(&graph->cmd_wait); graph->port = gpr_alloc_port(apm->gdev, dev, graph_callback, graph); - if (!graph->port) { + if (IS_ERR(graph->port)) { kfree(graph); - ret = -ENOMEM; + ret = PTR_ERR(graph->port); goto err; } -- 2.17.1

2 1

[v3] SoC: qcom: Distinguish headset codec by codec_dai->name
by Judy Hsiao 15 Dec '21

15 Dec '21

Distinguish which headset codec is on the board by codec_dai->name instead of card->name. It fixes the crash of being unable to handle kernel paging requests at virtual address ADDR by initializing the correct audio codec on the board. Call stack of the crash: ``` Unable to handle kernel paging request at virtual address ... ... Call trace: rt5682_set_component_pll+0xcc/0xb78 [snd_soc_rt5682] snd_soc_component_set_pll+0x90/0x154 snd_soc_dai_set_pll+0xf4/0x1ac sc7180_snd_startup+0x268/0x3c0 [snd_soc_sc7180] snd_soc_link_startup+0xa4/0x180 soc_pcm_open+0x35c/0x15c8 snd_pcm_open_substream+0xa90/0x13b0 snd_pcm_open+0x1a4/0x55c snd_pcm_capture_open+0x7c/0xe8 snd_open+0x2b8/0x2e4 chrdev_open+0x364/0x3d4 do_dentry_open+0x66c/0xc58 vfs_open+0x7c/0x8c path_openat+0x108c/0x2bbc do_filp_open+0x15c/0x258 do_sys_open+0x278/0x62c __arm64_compat_sys_openat+0x9c/0xb0 ... ``` Fixes: 425c5fce8a03 ("ASoC: qcom: Add support for ALC5682I-VS codec") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/qcom/sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 2fff764a00a7..37225ef2563a 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -131,13 +131,13 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int pll_id, pll_source, pll_in, pll_out, clk_id, ret; - if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + if (!strcmp(codec_dai->name, "rt5682-aif1")) { pll_source = RT5682_PLL1_S_MCLK; pll_id = 0; clk_id = RT5682_SCLK_S_PLL1; pll_out = RT5682_PLL1_FREQ; pll_in = DEFAULT_MCLK_RATE; - } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + } else if (!strcmp(codec_dai->name, "rt5682s-aif1")) { pll_source = RT5682S_PLL_S_MCLK; pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; -- 2.31.0

2 2

[PATCH] ASoC: rt5682: fix the wrong jack type detected
by derek.fang＠realtek.com 15 Dec '21

15 Dec '21

From: Derek Fang <derek.fang(a)realtek.com> Some powers were changed during the jack insert detection and clk's enable/disable in CCF. If in parallel, the influence has a chance to detect the wrong jack type, so add a lock. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 5224123d0d3b..b34a8542077d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -929,6 +929,8 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) unsigned int val, count; if (jack_insert) { + snd_soc_dapm_mutex_lock(dapm); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); @@ -979,6 +981,8 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) snd_soc_component_update_bits(component, RT5682_MICBIAS_2, RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK, RT5682_PWR_CLK25M_PU | RT5682_PWR_CLK1M_PU); + + snd_soc_dapm_mutex_unlock(dapm); } else { rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, -- 2.17.1

2 1

[PATCH v5 0/5] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 15 Dec '21

15 Dec '21

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V4: -- Update commit message and description of the chip specific extraction patch. -- Sort macros in kconfig and makefile. -- Update optional clock voting to conditional clock voting. -- Fix typo errors. -- Move to quicinc domain email id's. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (5): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Extract chip specific LPASS LPI code pinctrl: qcom: Add SC7280 lpass pin configuration pinctrl: qcom: Update clock voting as optional .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 130 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 130 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 249 ++------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 85 +++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 170 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 165 ++++++++++++++ 9 files changed, 697 insertions(+), 365 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- 2.7.4

5 16

[PATCH] ASoC: wcd934x: Fix a incorrect use of kstrndup
by Miaoqian Lin 14 Dec '21

14 Dec '21

In wcd934x_codec_enable_dec(), widget_name is allocated by kstrndup(). However, according to doc: "Note: Use kmemdup_nul() instead if the size is known exactly." So we should use kmemdup_nul() here instead of kstrndup(). It's similar to CVE-2019-12454. Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index e63c6b723d76..c6677cfbce59 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -5005,7 +5005,7 @@ static int wcd934x_codec_enable_dec(struct snd_soc_dapm_widget *w, char *dec; u8 hpf_coff_freq; - widget_name = kstrndup(w->name, 15, GFP_KERNEL); + widget_name = kmemdup_nul(w->name, 15, GFP_KERNEL); if (!widget_name) return -ENOMEM; -- 2.17.1

2 2

Re: linux-next: Tree for Dec 13 (SND_AMD_ACP_CONFIG)
by Randy Dunlap 14 Dec '21

14 Dec '21

On 12/13/21 19:02, broonie(a)kernel.org wrote: > Hi all, > > Non-merge commits (relative to Linus' tree): 5960 > 6555 files changed, 277265 insertions(+), 120864 deletions(-) > > ---------------------------------------------------------------------------- > on i386 or x86_64: when # CONFIG_ACPI is not set, so SND_SOC_ACPI is not set: WARNING: unmet direct dependencies detected for SND_AMD_ACP_CONFIG Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_ACPI [=n] Selected by [y]: - SND_SOC_AMD_ACP_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && X86 [=y] && PCI [=y] Full randconfig file is attached -- ~Randy

3 5

[PATCH 0/5] ASoC: qcom: apq8016_sbc: Allow routing audio through QDSP6
by Stephan Gerhold 14 Dec '21

14 Dec '21

This series makes it possible to route audio through the combined audio/modem DSP on MSM8916/APQ8016 devices instead of bypassing it using the LPASS drivers. This is necessary to support certain functionality such as voice call audio. See PATCH 4/5 for details. Also, qcom,apq8016-sbc.txt is converted to DT schema by adding it to the existing qcom,sm8250.yaml. The bindings are similar enough that it is easier to share a single schema instead of duplicating everything into multiple ones. Stephan Gerhold (5): ASoC: dt-bindings: qcom: sm8250: Drop redundant MultiMedia routes ASoC: dt-bindings: qcom: sm8250: Document "aux-devs" ASoC: dt-bindings: qcom: apq8016-sbc: Move to qcom,sm8250 DT schema ASoC: dt-bindings: qcom: Document qcom,msm8916-qdsp6-sndcard compatible ASoC: qcom: apq8016_sbc: Allow routing audio through QDSP6 .../bindings/sound/qcom,apq8016-sbc.txt | 96 ------------- .../bindings/sound/qcom,sm8250.yaml | 136 +++++++++++++++++- sound/soc/qcom/apq8016_sbc.c | 134 ++++++++++++++++- 3 files changed, 259 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt -- 2.34.1

4 16

[PATCH] ASoC: SOF: sof-probes: Constify sof_probe_compr_ops
by Rikard Falkeborn 14 Dec '21

14 Dec '21

The only usage of sof_probe_compr_ops is to assign its address to the cops field in the snd_soc_dai_driver struct (in sound/soc/sof/intel/hda-dai.c). Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/sof/sof-probes.c | 2 +- sound/soc/sof/sof-probes.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-probes.c b/sound/soc/sof/sof-probes.c index 5586af9f1a25..c79026cdb8c7 100644 --- a/sound/soc/sof/sof-probes.c +++ b/sound/soc/sof/sof-probes.c @@ -321,7 +321,7 @@ static int sof_probe_compr_pointer(struct snd_compr_stream *cstream, return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai); } -struct snd_soc_cdai_ops sof_probe_compr_ops = { +const struct snd_soc_cdai_ops sof_probe_compr_ops = { .startup = sof_probe_compr_startup, .shutdown = sof_probe_compr_shutdown, .set_params = sof_probe_compr_set_params, diff --git a/sound/soc/sof/sof-probes.h b/sound/soc/sof/sof-probes.h index 35e1dd8d9e03..4a1ed2942d28 100644 --- a/sound/soc/sof/sof-probes.h +++ b/sound/soc/sof/sof-probes.h @@ -32,7 +32,7 @@ int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, unsigned int *buffer_id, size_t num_buffer_id); -extern struct snd_soc_cdai_ops sof_probe_compr_ops; +extern const struct snd_soc_cdai_ops sof_probe_compr_ops; extern const struct snd_compress_ops sof_probe_compressed_ops; #endif -- 2.34.1

2 1

[PATCH v2] ASoC: SOF: Remove pm_runtime_put_autosuspend() for SOF OF device
by Daniel Baluta 14 Dec '21

14 Dec '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> In SOF OF device, pm_runtime_put_autosuspend() is not matching any pm_runtime_get_sync(). This is imbalanced for PM runtime. Also, for consistency we call pm_runtime_mark_last_busy() before enabling PM runtime. 1. Remove pm_runtime_put_autosuspend() in probe_complete 2. Reorder PM runtime calls int probe_complete Signed-off-by: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: - Added my own signed-off-by sound/soc/sof/sof-of-dev.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 412cbb824b84..b0089698eecb 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -74,11 +74,9 @@ static void sof_of_probe_complete(struct device *dev) /* allow runtime_pm */ pm_runtime_set_autosuspend_delay(dev, SND_SOF_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); pm_runtime_set_active(dev); pm_runtime_enable(dev); - - pm_runtime_mark_last_busy(dev); - pm_runtime_put_autosuspend(dev); } static int sof_of_probe(struct platform_device *pdev) -- 2.27.0

2 1

[PATCH] ASoC: mediatek: assign correct type to argument
by Jiaxin Yu 14 Dec '21

14 Dec '21

Fix the following sparse warning: (new ones prefixed by >>) >> sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c:370:33: sparse: sparse: incorrect type in argument 3 (different base types) sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c:370:33: sparse: expected unsigned int to sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c:370:33: sparse: got restricted snd_pcm_format_t [usertype] Correct discription of format, use S32_LE and S24_LE to distinguish the different 32bit. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 ++++---- .../mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 ++++++------ .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 ++-- .../mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c | 4 ++-- .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 4 ++-- 5 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index a4d26a6fc849..f8a72a5102ad 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -155,9 +155,9 @@ static const struct snd_soc_ops mt8183_da7219_rt1015_i2s_ops = { static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S32_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S32_LE); @@ -167,9 +167,9 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8183_rt1015_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index aeb1af86047e..d5fc86132b49 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -94,11 +94,11 @@ static const struct snd_soc_ops mt8183_mt6358_rt1015_i2s_ops = { static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - dev_dbg(rtd->dev, "%s(), fix format to 32bit\n", __func__); + dev_dbg(rtd->dev, "%s(), fix format to S32_LE\n", __func__); - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S32_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S32_LE); return 0; @@ -107,11 +107,11 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8183_rt1015_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - dev_dbg(rtd->dev, "%s(), fix format to 32bit\n", __func__); + dev_dbg(rtd->dev, "%s(), fix format to S24_LE\n", __func__); - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index a606133951b7..1d16939f80e3 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -350,9 +350,9 @@ static int mt8192_mt6359_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int mt8192_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c index cca1c739e690..5cdbfaafd479 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c @@ -359,7 +359,7 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) static int mt8195_etdm_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); @@ -464,7 +464,7 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index 11a185da0d96..fa50a31e9718 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -355,7 +355,7 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) static int mt8195_etdm_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); @@ -463,7 +463,7 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); -- 2.25.1

4 3

Re: [PATCH] ALSA: hda/realtek: Add new alc285-hp-amp-init model
by Takashi Iwai 14 Dec '21

14 Dec '21

On Mon, 13 Dec 2021 17:22:47 +0100, Bradley Scott wrote: > > Adds a new "alc285-hp-amp-init" model that can be used to apply the ALC285 > HP speaker amplifier initialization fixup to devices that are not already > known by passing "hda_model=alc285-hp-amp-init" to the > snd-sof-intel-hda-common module or "model=alc285-hp-amp-init" to the > snd-hda-intel module, depending on which is being used. > > Signed-off-by: Bradley Scott <bscott(a)teksavvy.com> Thanks, applied now. Takashi

1 0

[PATCH v2] ALSA: hda/realtek: Amp init fixup for HP ZBook 15 G6
by Bradley Scott 14 Dec '21

14 Dec '21

HP ZBook 15 G6 (SSID 103c:860f) needs the same speaker amplifier initialization as used on several other HP laptops using ALC285. Signed-off-by: Bradley Scott <Bradley.Scott(a)zebra.com> --- 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 3599f4c85ebf..d162662fe684 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8660,6 +8660,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), + SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), -- 2.25.1

2 1

[PATCH v4] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 14 Dec '21

14 Dec '21

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains', '#sound-dai-cells' and 'port' which were missing from txt file. Removes properties 'sram-size-playback' and 'sram-size-capture' since they are not used. Adds 'dmas' and 'dma-names' in the example which were missing from the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Changes 'op-mode', 'serial-dir' and 'tdm-slots' to optional properties as they are not needed if the McASP is used only as GPIO. Adds the yaml file in the 'MAINTAINERS' under the heading 'TEXAS INSTRUMENTS ASoC DRIVERS' Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- Changelog: v4: - changes the commit message - adds type and description to relevant properties - changes maxItems for 'serial-dir' - removes properties 'sram-size-playback' and 'sram-size-capture' as they are not used - removes 'function-gpios' - removes 'num-serializer' - marks 'tdm-slots', 'serial-dir' and 'op-mode' as optional properties - adds the yaml file in MAINTAINERS v3: - removes maxItems from 'clock-names' v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'ports' and 'num-serializer' as node properties .../bindings/sound/davinci-mcasp-audio.txt | 86 -------- .../bindings/sound/davinci-mcasp-audio.yaml | 185 ++++++++++++++++++ MAINTAINERS | 1 + 3 files changed, 186 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..5b553c90771a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,185 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: main registers + - description: data port register + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: I2S - 0 or DIT - 1 operation mode + enum: + - 0 + - 1 + + tdm-slots: + $ref: /schemas/types.yaml#/definitions/uint32 + description: number of channels over one serializer + maxItems: 1 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + All AXR pins should be present in the array even if inactive + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 25 + items: + minimum: 0 + maximum: 2 + default: 0 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: configures WFIFO threshold + maxItems: 1 + + rx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: configures RFIFO threshold + maxItems: 1 + + dismod: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + specify the drive on TX pin during inactive time slots + 0 - 3-state, 2 - logic low, 3 - logic high + enum: + - 0 + - 2 + - 3 + default: 2 + + interrupts: + anyOf: + - minItems: 1 + items: + - description: TX FIFO interrupt + - description: RX FIFO interrupt + - items: + - description: common FIFO interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + $ref: /schemas/types.yaml#/definitions/string + description: parent clock name for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + $ref: /schemas/types.yaml#/definitions/uint32 + description: ratio of AUCLK and FS rate if applicable + maxItems: 1 + + gpio-controller: true + + "#gpio-cells": + const: 2 + + clocks: + minItems: 1 + items: + - description: functional clock + - description: module specific optional ahclkx clock + - description: module specific optional ahclkr clock + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + description: phandle to the corresponding power-domain + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + port: + description: connection for when McASP is used via graph card + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 00ad0cb5cb05..3fdf1e23f7d7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18855,6 +18855,7 @@ TEXAS INSTRUMENTS ASoC DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ TEXAS INSTRUMENTS' DAC7612 DAC DRIVER -- 2.17.1

3 7

Does alsa-lib call this kernel api?
by GitHub issues - opened 14 Dec '21

14 Dec '21

alsa-project/alsa-lib issue #204 was opened from Nick0412: Where in the alsa library does it reference the alsa kernel api: https://www.kernel.org/doc/html/latest/sound/kernel-api/alsa-driver-api.htm… Do these two apis interact at all? Issue URL : https://github.com/alsa-project/alsa-lib/issues/204 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/4] fsl-asoc-card: Add optional dt property for setting mclk-id
by Ariel D'Alessandro 14 Dec '21

14 Dec '21

This is a follow up of patchset: [RFC patch 0/5] Support BCLK input clock in tlv320aic31xx Sound cards may allow using different main clock inputs. In the generic fsl-asoc-card driver, these values are hardcoded for each specific card configuration. Let's make it more flexible, allowing setting mclk-id from the device-tree node. Ariel D'Alessandro (4): dt-bindings: sound: Rename tlv320aic31xx-micbias as tlv320aic31xx dt-bindings: tlv320aic31xx: Define PLL clock inputs ASoC: fsl-asoc-card: Add optional dt property for setting mclk-id ASoC: fsl-asoc-card: Remove BCLK default value for tlv320aic31xx card .../devicetree/bindings/sound/fsl-asoc-card.txt | 1 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 2 +- arch/arm/boot/dts/am43x-epos-evm.dts | 2 +- include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 --------- include/dt-bindings/sound/tlv320aic31xx.h | 14 ++++++++++++++ sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 7 ++++++- 7 files changed, 24 insertions(+), 13 deletions(-) delete mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 include/dt-bindings/sound/tlv320aic31xx.h -- 2.30.2

2 9

[PATCH 0/7] ASoC: Intel: SOF: clarifications on hardware support
by Pierre-Louis Bossart 14 Dec '21

14 Dec '21

This patchset revisits the Intel hardware support in SOF. The HDAudio DMA position information was not following hardware recommended programming sequences (similar changes are already part of the HDaudio legacy driver), and the stream assignment applied a work-around that was only needed on specific versions of hardware. These changes are not tagged as 'Fixes' and don't need to be applied to -stable versions. While we're at it, DPIB handling is improved, useless fields removed, a comment added on JasperLake support, and IceLake-specific routines are isolated. Pierre-Louis Bossart (6): ASoC: SOF: Intel: hda-stream: limit PROCEN workaround ASoC: SOF: Intel: hda-ctrl: apply symmetry for DPIB ASoC: SOF: hda-stream: only enable DPIB if needed ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information ASoC: SOF: Intel: hda-dai: remove unused fields ASoC: SOF: Intel: add comment on JasperLake support Ranjani Sridharan (1): ASoC: SOF: Intel: ICL: move ICL-specific ops to icl.c sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 7 +++ sound/soc/sof/intel/hda-ctrl.c | 2 +- sound/soc/sof/intel/hda-dai.c | 4 -- sound/soc/sof/intel/hda-loader.c | 64 ------------------------ sound/soc/sof/intel/hda-pcm.c | 86 ++++++++++++++++++++++---------- sound/soc/sof/intel/hda-stream.c | 25 ++++++---- sound/soc/sof/intel/hda.c | 9 +++- sound/soc/sof/intel/hda.h | 8 ++- sound/soc/sof/intel/icl.c | 67 ++++++++++++++++++++++++- sound/soc/sof/intel/shim.h | 4 ++ 11 files changed, 169 insertions(+), 108 deletions(-) -- 2.25.1

2 8

[PATCH] ASoC: dt-bindings: tegra: Document interconnects property
by Thierry Reding 14 Dec '21

14 Dec '21

From: Thierry Reding <treding(a)nvidia.com> Add the interconnects and interconnect-names properties to the bindings for the sound card on various NVIDIA Tegra based boards. These are used to describe the device's memory paths to and from memory. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- .../bindings/sound/nvidia,tegra-audio-graph-card.yaml | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml index 5bdd30a8a404..b4bee466d67a 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml @@ -44,6 +44,16 @@ properties: minItems: 1 maxItems: 3 + interconnects: + items: + - description: APE read memory client + - description: APE write memory client + + interconnect-names: + items: + - const: dma-mem # read + - const: write + iommus: maxItems: 1 -- 2.33.1

3 2

[PATCH v1 1/2] ASoC: tegra: Add DAPM switches for headphones and mic jack
by Dmitry Osipenko 14 Dec '21

14 Dec '21

UCM of Acer Chromebook (Nyan) uses DAPM switches of headphones and mic jack. These switches were lost by accident during unification of the machine drivers, restore them. Cc: <stable(a)vger.kernel.org> Fixes: cc8f70f ("ASoC: tegra: Unify ASoC machine drivers") Reported-by: Thomas Graichen <thomas.graichen(a)gmail.com> # T124 Nyan Big Tested-by: Thomas Graichen <thomas.graichen(a)gmail.com> # T124 Nyan Big Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- sound/soc/tegra/tegra_asoc_machine.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index b95438c3dbf7..f3e86bd714b4 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -116,6 +116,8 @@ static const struct snd_kcontrol_new tegra_machine_controls[] = { SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Internal Mic 1"), SOC_DAPM_PIN_SWITCH("Internal Mic 2"), + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), }; int tegra_asoc_machine_init(struct snd_soc_pcm_runtime *rtd) -- 2.33.1

2 2

[PATCH] SoC: qcom: Distinguish headset codec by codec_dai->name
by Judy Hsiao 13 Dec '21

13 Dec '21

Distinguish which headset codec is on the board by codec_dai->name instead of card->name. Fixes: 425c5fce8a03 ("ASoC: qcom: Add support for ALC5682I-VS codec") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/qcom/sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 2fff764a00a7..37225ef2563a 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -131,13 +131,13 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int pll_id, pll_source, pll_in, pll_out, clk_id, ret; - if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + if (!strcmp(codec_dai->name, "rt5682-aif1")) { pll_source = RT5682_PLL1_S_MCLK; pll_id = 0; clk_id = RT5682_SCLK_S_PLL1; pll_out = RT5682_PLL1_FREQ; pll_in = DEFAULT_MCLK_RATE; - } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + } else if (!strcmp(codec_dai->name, "rt5682s-aif1")) { pll_source = RT5682S_PLL_S_MCLK; pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; -- 2.34.1.173.g76aa8bc2d0-goog

3 2

[PATCH] ASoC: dt-bindings: add missing pins
by David Heidelberg 13 Dec '21

13 Dec '21

Add pins missing in documentation, but present in the hardware. Fixes: 0e826e867264 ("ASoC: add RT5677 CODEC driver") Signed-off-by: David Heidelberg <david(a)ixit.cz> --- Documentation/devicetree/bindings/sound/rt5677.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index da2430099181..74b6359b802d 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -54,9 +54,21 @@ Pins on the device (for linking into audio routes): * DMIC2 * DMIC3 * DMIC4 + * DMIC L1 + * DMIC L2 + * DMIC L3 + * DMIC L4 + * DMIC R1 + * DMIC R2 + * DMIC R3 + * DMIC R4 * LOUT1 * LOUT2 * LOUT3 + * PDM1L + * PDM1R + * PDM2L + * PDM2R Example: -- 2.33.0

2 1

Re: [PATCH v2 1/4] ASoC: dt-bindings: stm32: i2s: add audio-graph-card port
by Rob Herring 13 Dec '21

13 Dec '21

On Tue, Dec 7, 2021 at 7:52 AM Alexandre TORGUE <alexandre.torgue(a)foss.st.com> wrote: > > Hi Rob > > On 12/1/21 11:34 PM, Rob Herring wrote: > > On Fri, Nov 26, 2021 at 11:25:27AM +0100, Olivier MOYSAN wrote: > >> Hi Rob, > >> > >> On 11/25/21 10:26 PM, Rob Herring wrote: > >>> On Thu, 25 Nov 2021 15:40:50 +0100, Olivier Moysan wrote: > >>>> The STM2 I2S DAI can be connected via the audio-graph-card. > >>>> Add port entry into the bindings. > >>>> > >>>> Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> > >>>> --- > >>>> Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 5 +++++ > >>>> 1 file changed, 5 insertions(+) > >>>> > >>> > >>> Running 'make dtbs_check' with the schema in this patch gives the > >>> following warnings. Consider if they are expected or the schema is > >>> incorrect. These may not be new warnings. > >>> > >>> Note that it is not yet a requirement to have 0 warnings for dtbs_check. > >>> This will change in the future. > >>> > >>> Full log is available here: https://patchwork.ozlabs.org/patch/1559750 > >>> > >>> > >>> audio-controller@4000b000: 'port' does not match any of the regexes: '^port@[0-9]', 'pinctrl-[0-9]+' > >>> arch/arm/boot/dts/stm32mp157a-dk1.dt.yaml > >>> arch/arm/boot/dts/stm32mp157c-dk2.dt.yaml > >>> > >> > >> This warning is not a new one. > >> > >> The i2s2 node in stm32mp15xx-dkx.dtsi would require the following binding: > >> port: > >> $ref: audio-graph-port.yaml# > >> unevaluatedProperties: false > >> > >> However the spi binding requires to introduce a unit address: > >> patternProperties: > >> '^port@[0-9]': > >> $ref: audio-graph-port.yaml# > >> unevaluatedProperties: false > >> > >> The warning can be removed by re-ordering the bindings patches in the serie, > >> as "additionalProperties: true" makes the check more tolerant on extra > >> properties. > > > > That's never right. > > > >> The patch "ASoC: dt-bindings: stm32: i2s: add audio-graph-card port" can > >> even be merely dropped. > >> So, I suggest to resend the serie without audio-graph-card patch. > > > > Only if you aren't using audio-graph-card. > > > >> > >> Does it sound too permissive to you ? > > > > I think perhaps you need to combine the schemas into 1. Or you need to > > restructure your dtsi files such that you only add spi specific > > properties when spi mode is enabled and only add i2s specific properties > > when i2s mode is enabled. Or use the /delete-property/ directive. > > Initially the aim of this series was to fix a "make W=1" warnings seen > on spi and i2s nodes (duplicate unit-address). Moving both nodes in a > common node + using a different compatible depending on SPI or I2S usage > sounded good) but it is not enough. In this series the common node is > named as following: "spi2s2: spi@4000b000". It is fine for a spi usage > but if we want to use this "common node" with I2S compatible and > specific bindings, the node name remains spi@... and then specific spi > checks are done. For this with this series applied we got this issue > reported by spi-controller.yaml: > > spi@4000b000: port@0: 'compatible' is a required property > > So, if we use two separates nodes we got W=1 warning and if we use a > common node we got yaml check issue. One possibility would be to use a > common node with a new node name (for example i2spi@...) but I think it > is not acceptable. It is acceptable, see this thread[1]. Rob [1] https://lore.kernel.org/all/20211203183517.11390-1-semen.protsenko@linaro.o…

2 1

[v2] SoC: qcom: Distinguish headset codec by codec_dai->name
by Judy Hsiao 13 Dec '21

13 Dec '21

Distinguish which headset codec is on the board by codec_dai->name instead of card->name. It fixes the crash of being unable to handle kernel paging requests at virtual address ADDR by initializing the correct audio codec on the board. Call stack of the crash: ``` Unable to handle kernel paging request at virtual address ffffffbfe7bba9ce Mem abort info: ESR = 0x96000005 EC = 0x25: DABT (current EL), IL = 32 bits SET = 0, FnV = 0 EA = 0, S1PTW = 0 Data abort info: ISV = 0, ISS = 0x00000005 CM = 0, WnR = 0 swapper pgtable: 4k pages, 39-bit VAs, pgdp=000000008379a000 [ffffffbfe7bba9ce] pgd=0000000000000000, pud=0000000000000000 Internal error: Oops: 96000005 [#1] PREEMPT SMP Modules linked in: ip6table_nat fuse uvcvideo videobuf2_vmalloc [trimmed] CPU: 7 PID: 2168 Comm: cras Not tainted 5.4.163-lockdep-17364-gfe3d4f499cf1 #1 Hardware name: Google Pompom (rev2) with LTE (DT) pstate: 00400009 (nzcv daif +PAN -UAO) pc : rt5682_set_component_pll+0xcc/0xb78 [snd_soc_rt5682] lr : rt5682_set_component_pll+0xbc/0xb78 [snd_soc_rt5682] sp : ffffff808ed7f320 x29: ffffff808ed7f390 x28: dfffffd000000000 x27: ffffff80b1409550 x26: 00000000aaaaaaaa x25: ffffff80d0a0b820 x24: ffffff80bc1f7098 x23: ffffff809332a080 x22: 00000000aaaaaaaa x21: 1ffffff01783ee13 x20: 00000000aaaaaaaa x19: 00000000aaaaaaaa x18: 1ffffff011dafe18 x17: 0000000000000000 x16: 0000000000000201 x15: 0000000000000000 x14: 0000000062f77d15 x13: dfffffd000000000 x12: ffffffd01302ed7c x11: 0000000000000000 x10: ffffff7f3ddd4e74 x9 : 0000000000000000 x8 : 1fffffefe7bba9ce x7 : aaaaaaaaaaaaaaaa x6 : 0000000000000000 x5 : 0000000000000000 x4 : 0000000000000008 x3 : 0000000000000000 x2 : 0000000000000008 x1 : 00000000000000aa x0 : ffffff808ed7f358 Call trace: rt5682_set_component_pll+0xcc/0xb78 [snd_soc_rt5682] snd_soc_component_set_pll+0x90/0x154 snd_soc_dai_set_pll+0xf4/0x1ac sc7180_snd_startup+0x268/0x3c0 [snd_soc_sc7180] snd_soc_link_startup+0xa4/0x180 soc_pcm_open+0x35c/0x15c8 snd_pcm_open_substream+0xa90/0x13b0 snd_pcm_open+0x1a4/0x55c snd_pcm_capture_open+0x7c/0xe8 snd_open+0x2b8/0x2e4 chrdev_open+0x364/0x3d4 do_dentry_open+0x66c/0xc58 vfs_open+0x7c/0x8c path_openat+0x108c/0x2bbc do_filp_open+0x15c/0x258 do_sys_open+0x278/0x62c __arm64_compat_sys_openat+0x9c/0xb0 el0_svc_common+0x1c0/0x3dc el0_svc_compat_handler+0x88/0xd4 el0_svc_compat+0x8/0x2c Code: 8b3acae8 910d310a d343fd48 a9012be8 (38fc6908) ``` Fixes: 425c5fce8a03 ("ASoC: qcom: Add support for ALC5682I-VS codec") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/qcom/sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 2fff764a00a7..37225ef2563a 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -131,13 +131,13 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int pll_id, pll_source, pll_in, pll_out, clk_id, ret; - if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + if (!strcmp(codec_dai->name, "rt5682-aif1")) { pll_source = RT5682_PLL1_S_MCLK; pll_id = 0; clk_id = RT5682_SCLK_S_PLL1; pll_out = RT5682_PLL1_FREQ; pll_in = DEFAULT_MCLK_RATE; - } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + } else if (!strcmp(codec_dai->name, "rt5682s-aif1")) { pll_source = RT5682S_PLL_S_MCLK; pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; -- 2.31.0

2 1

[PATCH v2] ALSA: gus: Fix memory leaks at memory allocator error paths
by Takashi Iwai 13 Dec '21

13 Dec '21

When snd_gf1_mem_xalloc() returns NULL, the current code still leaves the formerly allocated block.name string but returns an error immediately. This patch does code-refactoring to move the kstrdup() call itself into snd_gf1_mem_xalloc() and deals with the resource free in the helper code by itself for fixing those memory leaks. Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20211213132444.22385-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/gus/gus_mem.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 5e3ff3137dd7..3e56c01c4544 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -24,8 +24,9 @@ void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup) } } -static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, - struct snd_gf1_mem_block * block) +static struct snd_gf1_mem_block * +snd_gf1_mem_xalloc(struct snd_gf1_mem *alloc, struct snd_gf1_mem_block *block, + const char *name) { struct snd_gf1_mem_block *pblock, *nblock; @@ -33,6 +34,12 @@ static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, if (nblock == NULL) return NULL; *nblock = *block; + nblock->name = kstrdup(name, GFP_KERNEL); + if (!nblock->name) { + kfree(nblock); + return NULL; + } + pblock = alloc->first; while (pblock) { if (pblock->ptr > nblock->ptr) { @@ -198,12 +205,7 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne if (share_id != NULL) memcpy(&block.share_id, share_id, sizeof(block.share_id)); block.owner = owner; - block.name = kstrdup(name, GFP_KERNEL); - if (block.name == NULL) { - snd_gf1_mem_lock(alloc, 1); - return NULL; - } - nblock = snd_gf1_mem_xalloc(alloc, &block); + nblock = snd_gf1_mem_xalloc(alloc, &block, name); snd_gf1_mem_lock(alloc, 1); return nblock; } @@ -240,14 +242,12 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) if (gus->gf1.enh_mode) { block.ptr = 0; block.size = 1024; - block.name = kstrdup("InterWave LFOs", GFP_KERNEL); - if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (!snd_gf1_mem_xalloc(alloc, &block, "InterWave LFOs")) return -ENOMEM; } block.ptr = gus->gf1.default_voice_address; block.size = 4; - block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL); - if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (!snd_gf1_mem_xalloc(alloc, &block, "Voice default (NULL's)")) return -ENOMEM; #ifdef CONFIG_SND_DEBUG snd_card_ro_proc_new(gus->card, "gusmem", gus, snd_gf1_mem_info_read); -- 2.31.1

2 1

alsa crashing each time my screen gets locked
by GitHub issues - opened 13 Dec '21

13 Dec '21

alsa-project/alsa-lib issue #203 was opened from Fallast: each time i get my screen lock after being inactive for 5 minutes i log back in and alsa stops working and my volume icon disappears i have to run volumeicon to get it working back again i'm on mx linux, based on debian bullseye Issue URL : https://github.com/alsa-project/alsa-lib/issues/203 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ALSA: gus: Fix erroneous memory allocation
by Takashi Iwai 13 Dec '21

13 Dec '21

snd_gf1_mem_xalloc() returns NULL incorrectly when the memory chunk is allocated in the middle of the chain. This patch corrects the return value to treat it properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/gus/gus_mem.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 4c691dbf2721..5e3ff3137dd7 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -44,7 +44,7 @@ static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, else nblock->prev->next = nblock; mutex_unlock(&alloc->memory_mutex); - return NULL; + return nblock; } pblock = pblock->next; } -- 2.31.1

2 4

[PATCH v2] ALSA: sound/isa/gus: check the return value of kstrdup()
by Xiaoke Wang 13 Dec '21

13 Dec '21

kstrdup() returns NULL when some internal memory errors happen, it is better to check the return value of it. Otherwise, we may not to be able to catch some memory errors in time. Signed-off-by: Xiaoke Wang <xkernel.wang(a)foxmail.com> --- Changes in v2: - add the proper error handling. Thanks for the suggestions from Takashi Iwai. In the future, I'll pay more attention to what he mentioned. --- sound/isa/gus/gus_mem.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index ff9480f..4c691db 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -199,6 +199,10 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne memcpy(&block.share_id, share_id, sizeof(block.share_id)); block.owner = owner; block.name = kstrdup(name, GFP_KERNEL); + if (block.name == NULL) { + snd_gf1_mem_lock(alloc, 1); + return NULL; + } nblock = snd_gf1_mem_xalloc(alloc, &block); snd_gf1_mem_lock(alloc, 1); return nblock; @@ -237,13 +241,13 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) block.ptr = 0; block.size = 1024; block.name = kstrdup("InterWave LFOs", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; } block.ptr = gus->gf1.default_voice_address; block.size = 4; block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; #ifdef CONFIG_SND_DEBUG snd_card_ro_proc_new(gus->card, "gusmem", gus, snd_gf1_mem_info_read); --

2 1

Re: [PATCH] ALSA: sound/isa/gus: check the return value of kstrdup()
by Xiaoke Wang 13 Dec '21

13 Dec '21

Note: Compare with the last email, this one is using my full name. kstrdup() returns NULL when some internal memory errors happen, it is better to check the return value of it. Otherwise, we may not to be able to catch some memory errors in time. Signed-off-by: Xiaoke Wang <xkernel.wang(a)foxmail.com> --- sound/isa/gus/gus_mem.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index ff9480f..f8d915f 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -199,6 +199,8 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne memcpy(&block.share_id, share_id, sizeof(block.share_id)); block.owner = owner; block.name = kstrdup(name, GFP_KERNEL); + if (block.name == NULL) + return NULL; nblock = snd_gf1_mem_xalloc(alloc, &block); snd_gf1_mem_lock(alloc, 1); return nblock; @@ -237,13 +239,13 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) block.ptr = 0; block.size = 1024; block.name = kstrdup("InterWave LFOs", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; } block.ptr = gus->gf1.default_voice_address; block.size = 4; block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; #ifdef CONFIG_SND_DEBUG snd_card_ro_proc_new(gus->card, "gusmem", gus, snd_gf1_mem_info_read); --

2 1

Re: [PATCH] ALSA: jack: Check the return value of kstrdup()
by Xiaoke Wang 13 Dec '21

13 Dec '21

kstrdup() can return NULL, it is better to check the return value of it. Fix: free jack and use my full name. Signed-off-by: Xiaoke Wang <xkernel.wang(a)foxmail.com> --- sound/core/jack.c | 4 ++ 1 file changed, 4 insertions(+) diff --git a/sound/core/jack.c b/sound/core/jack.c index 32350c6..5fd9954 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -509,6 +509,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; jack->id = kstrdup(id, GFP_KERNEL); + if (jack->id == NULL) { + kfree(jack); + return -ENOMEM; + } /* don't creat input device for phantom jack */ if (!phantom_jack) { --

2 1

[PATCH] ALSA: drivers: opl3: Fix incorrect use of vp->state
by Colin Ian King 13 Dec '21

13 Dec '21

Static analysis with scan-build has found an assignment to vp2 that is never used. It seems that the check on vp->state > 0 should be actually on vp2->state instead. Fix this. This dates back to 2002, I found the offending commit from the git history git://git.kernel.org/pub/scm/linux/kernel/git/tglx/history.git, commit 91e39521bbf6 ("[PATCH] ALSA patch for 2.5.4") Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/drivers/opl3/opl3_midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index e1b69c65c3c8..e2b7be67f0e3 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -397,7 +397,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } if (instr_4op) { vp2 = &opl3->voices[voice + 3]; - if (vp->state > 0) { + if (vp2->state > 0) { opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset + 3); reg_val = vp->keyon_reg & ~OPL3_KEYON_BIT; -- 2.33.1

2 1

[PATCH] ALSA: uapi: use C90 comment style instead of C99 style
by Takashi Sakamoto 13 Dec '21

13 Dec '21

UAPI headers are built with compiler option for C90, thus double-slashes comment introduced in C99 is not preferable. Fixes: fb6723daf890 ("ALSA: pcm: comment about relation between msbits hw parameter and [S|U]32 formats") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/uapi/sound/asound.h | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index cf1d20e34167..1834f58b8ede 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -202,9 +202,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S24_BE ((__force snd_pcm_format_t) 7) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_LE ((__force snd_pcm_format_t) 8) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_BE ((__force snd_pcm_format_t) 9) /* low three bytes */ -// For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the -// available bit count in most significant bit. It's for the case of so-called 'left-justified' or -// `right-padding` sample which has less width than 32 bit. +/* + * For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the + * available bit count in most significant bit. It's for the case of so-called 'left-justified' or + * `right-padding` sample which has less width than 32 bit. + */ #define SNDRV_PCM_FORMAT_S32_LE ((__force snd_pcm_format_t) 10) #define SNDRV_PCM_FORMAT_S32_BE ((__force snd_pcm_format_t) 11) #define SNDRV_PCM_FORMAT_U32_LE ((__force snd_pcm_format_t) 12) -- 2.32.0

2 1

[PATCH] ALSA: pcm: comment about relation between msbits hw parameter and [S|U32] formats
by Takashi Sakamoto 13 Dec '21

13 Dec '21

Regarding to handling [U|S][32|24] PCM formats, many userspace application developers and driver developers have confusion, since they require them to understand justification or padding. It easily loses consistency and soundness to operate with many type of devices. In this commit, I attempt to solve the situation by adding comment about relation between [S|U]32 formats and 'msbits' hardware parameter. The formats are used for 'left-justified' sample format, and the available bit count in most significant bit is delivered to userspace in msbits hardware parameter (struct snd_pcm_hw_params.msbits), which is decided by msbits constraint added by pcm drivers (snd_pcm_hw_constraint_msbits()). In driver side, the msbits constraint includes two elements; the physical width of format and the available width of the format in most significant bit. The former is used to match SAMPLE_BITS of format. (For my convenience, I ignore wildcard in the usage of the constraint.) As a result of interaction between ALSA pcm core and ALSA pcm application, when the format in which SAMPLE_BITS equals to physical width of the msbits constaint, the msbits parameter is set by referring to the available width of the constraint. When the msbits parameter is not changed in the above process, ALSA pcm core set it alternatively with SAMPLE_BIT of chosen format. In userspace application side, the msbits is only available after calling ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS request. Even if the hardware parameter structure includes somewhat value of SAMPLE_BITS interval parameter as width of format, all of the width is not always available since msbits can be less than the width. I note that [S|U24] formats are used for 'right-justified' 24 bit sample formats within 32 bit frame. The first byte in most significant bit should be invalidated. Although the msbits exposed to userspace should be zero as invalid value, actually it is 32 from physical width of format. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/pcm.h | 3 +++ include/uapi/sound/asound.h | 3 +++ 2 files changed, 6 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2e1200d17d0c..b84ff6e4ee53 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -147,6 +147,9 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_S24_BE _SNDRV_PCM_FMTBIT(S24_BE) #define SNDRV_PCM_FMTBIT_U24_LE _SNDRV_PCM_FMTBIT(U24_LE) #define SNDRV_PCM_FMTBIT_U24_BE _SNDRV_PCM_FMTBIT(U24_BE) +// For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the +// available bit count in most significant bit. It's for the case of so-called 'left-justified' or +// `right-padding` sample which has less width than 32 bit. #define SNDRV_PCM_FMTBIT_S32_LE _SNDRV_PCM_FMTBIT(S32_LE) #define SNDRV_PCM_FMTBIT_S32_BE _SNDRV_PCM_FMTBIT(S32_BE) #define SNDRV_PCM_FMTBIT_U32_LE _SNDRV_PCM_FMTBIT(U32_LE) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index d17c061950df..b43ea0067e1f 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -202,6 +202,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S24_BE ((__force snd_pcm_format_t) 7) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_LE ((__force snd_pcm_format_t) 8) /* low three bytes */ #define SNDRV_PCM_FORMAT_U24_BE ((__force snd_pcm_format_t) 9) /* low three bytes */ +// For S32/U32 formats, 'msbits' hardware parameter is often used to deliver information about the +// available bit count in most significant bit. It's for the case of so-called 'left-justified' or +// `right-padding` sample which has less width than 32 bit. #define SNDRV_PCM_FORMAT_S32_LE ((__force snd_pcm_format_t) 10) #define SNDRV_PCM_FORMAT_S32_BE ((__force snd_pcm_format_t) 11) #define SNDRV_PCM_FORMAT_U32_LE ((__force snd_pcm_format_t) 12) -- 2.27.0

2 5

No audio from Starship/Matisse HD Audio Controller
by GitHub issues - edited 13 Dec '21

13 Dec '21

alsa-project/alsa-ucm-conf issue #129 was edited from SirFooOfBar: I have an AMD ryzen 3900X paired with a Gigabyte AX370-Gaming 5 motherboard. I use arch linux. Upon upgrading the package alsa-ucm-conf from version 1.2.6.1-1 to 1.2.6.2-2 and rebooting, my onboard audio stops working completely. Downgrading restores functionality after another reboot. In pavucontrol, for 1.2.6.1-1 there are many available profiles, but for 1.2.6.2-2 there is only one, which says unavailable. I also use pipewire rather than pulseaudio if that makes a difference Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/129 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No audio from Starship/Matisse HD Audio Controller
by GitHub issues - opened 13 Dec '21

13 Dec '21

alsa-project/alsa-ucm-conf issue #129 was opened from SirFooOfBar: I have an AMD ryzen 3900X paired with a Gigabyte AX370-Gaming 5 motherboard. I use arch linux. Upon upgrading the package alsa-ucm-conf from version 1.2.6.1-1 to 1.2.6.2-2 and rebooting, my onboard audio stops working completely. Downgrading restores functionality after another reboot. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/129 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: sound/isa/gus: check the return value of kstrdup()
by xkernel 12 Dec '21

12 Dec '21

kstrdup() returns NULL when some internal memory errors happen, it is better to check the return value of it. Otherwise, we may not to be able to catch some memory errors in time. Signed-off-by: xkernel <xkernel.wang(a)foxmail.com> --- sound/isa/gus/gus_mem.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index ff9480f..f8d915f 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -199,6 +199,8 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne memcpy(&block.share_id, share_id, sizeof(block.share_id)); block.owner = owner; block.name = kstrdup(name, GFP_KERNEL); + if (block.name == NULL) + return NULL; nblock = snd_gf1_mem_xalloc(alloc, &block); snd_gf1_mem_lock(alloc, 1); return nblock; @@ -237,13 +239,13 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) block.ptr = 0; block.size = 1024; block.name = kstrdup("InterWave LFOs", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; } block.ptr = gus->gf1.default_voice_address; block.size = 4; block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL); - if (snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; #ifdef CONFIG_SND_DEBUG snd_card_ro_proc_new(gus->card, "gusmem", gus, snd_gf1_mem_info_read); --

1 0

Move the call to expand variables and expressions
by GitHub pull_request - edited 12 Dec '21

12 Dec '21

alsa-project/alsa-utils pull request #127 was edited from ranj063: With the addition of the math expression evaluator in alsa-lib, we can use it to evaluate simple math expresionns for objects attribute values based on other attributes. For ex: we can set the expression for buffer size as follows: buffer_size "$[($in_channels * 48) * 4]" The buffer_size attribute value will be computed with the attribute value "in_channels" based on the expression above. So if $in_channels = 2, buffer_size will be evaluated to 384. Request URL : https://github.com/alsa-project/alsa-utils/pull/127 Patch URL : https://github.com/alsa-project/alsa-utils/pull/127.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Move the call to expand variables and expressions
by GitHub pull_request - edited 12 Dec '21

12 Dec '21

alsa-project/alsa-utils pull request #127 was edited from ranj063: With the addition of the math expression evaluator in alsa-lib, we can use it to evaluate simple math expresionns for objects attribute values based on other attributes. For ex: we can set the expression for buffer size as follows: buffer_size "$[$[$in_channels * 48] * 4]" The buffer_size attribute value will be computed with the attribute value "in_channels" based on the expression above. So if $in_channels = 2, buffer_size will be evaluated to 384. Request URL : https://github.com/alsa-project/alsa-utils/pull/127 Patch URL : https://github.com/alsa-project/alsa-utils/pull/127.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

S24LE and S32LE format equivalency
by Marios Sioutis 12 Dec '21

12 Dec '21

Hi all, Quick question, I was troubleshooting a 24bit USB device and got down the rabbit hole of sample formats. The device advertises itself as S24LE (4byte payload) but it appears as S32LE under alsa, something that bothered me a lot initially. However, If the conversion between the two formats is to just << 8 i.e. put a zero byte as LSB, then the two formats are essentially equivalent. Is this actually the case? If yes, who performs the conversion? Is it on the alsa side, or the device already adds a zero byte LSB? Thank you in advance, Marios

2 3

[PATCH v4 0/3] kselftest: alsa: Add basic mixer selftest
by Mark Brown 12 Dec '21

12 Dec '21

This series adds a basic selftest for the mixer interface, as discussed in the cover letter for the main patch there's plenty of additional coverage that could be added but this is a good starting point. v4: - More stylistic updates suggested by Shuah. - Fix build with older alsa-lib. v3: - Pull in incremental updates adding a fixed library configuration from Jaroslav and support for volatile controls from Takashi Sakamoto. - Stylistic updates suggested by Shuah. v2: - Use pkg-config to get CFLAGS and LDLIBS for alsa-lib. Jaroslav Kysela (1): kselftest: alsa: Use private alsa-lib configuration in mixer test Mark Brown (1): kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest Takashi Sakamoto (1): kselftest: alsa: optimization for SNDRV_CTL_ELEM_ACCESS_VOLATILE MAINTAINERS | 8 + tools/testing/selftests/Makefile | 3 +- tools/testing/selftests/alsa/.gitignore | 1 + tools/testing/selftests/alsa/Makefile | 9 + tools/testing/selftests/alsa/mixer-test.c | 665 ++++++++++++++++++++++ 5 files changed, 685 insertions(+), 1 deletion(-) create mode 100644 tools/testing/selftests/alsa/.gitignore create mode 100644 tools/testing/selftests/alsa/Makefile create mode 100644 tools/testing/selftests/alsa/mixer-test.c base-commit: fa55b7dcdc43c1aa1ba12bca9d2dd4318c2a0dbf -- 2.30.2

2 4

Add ALC287 speaker fixup for Lenovo IdeaPad Slim 9i 14ITL5
by Bart Kroon 12 Dec '21

12 Dec '21

Hello, The speaker fixup that is used for the Yoga 7 14ITL5 also applies to the IdeaPad Slim 9i 14ITL5. The attached patch applies the quirk. The attached patch is validated on my laptop. Regards, Bart

2 1

[PATCH] ALSA: sparc: no need to initialise statics to 0
by Jason Wang 12 Dec '21

12 Dec '21

Static variables do not need to be initialised to 0, because compiler will initialise all uninitialised statics to 0. Thus, remove the unneeded initializations. Signed-off-by: Jason Wang <wangborong(a)cdjrlc.com> --- sound/sparc/dbri.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 6b84f66e4af4..3881e1c1b08a 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -688,7 +688,7 @@ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) { u32 dvma_addr = (u32)dbri->dma_dvma; s32 tmp, addr; - static int wait_id = 0; + static int wait_id; wait_id++; wait_id &= 0xffff; /* restrict it to a 16 bit counter. */ @@ -1926,7 +1926,7 @@ static void dbri_process_interrupt_buffer(struct snd_dbri *dbri) static irqreturn_t snd_dbri_interrupt(int irq, void *dev_id) { struct snd_dbri *dbri = dev_id; - static int errcnt = 0; + static int errcnt; int x; if (dbri == NULL) @@ -2591,7 +2591,7 @@ static int dbri_probe(struct platform_device *op) struct snd_dbri *dbri; struct resource *rp; struct snd_card *card; - static int dev = 0; + static int dev; int irq; int err; -- 2.34.1

2 1

ALSA: drivers: opl3: assignment of a pointer that is not used, probable bug
by Colin King (gmail) 12 Dec '21

12 Dec '21

Hi, static analysis with scan-build has found an assignment to vp2 that is never used in function snd_opl3_note_on(), sound/drivers/opl3/opl3_midi.c as follows: if (instr_4op) { vp2 = &opl3->voices[voice + 3]; if (vp->state > 0) { opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset + 3); reg_val = vp->keyon_reg & ~OPL3_KEYON_BIT; opl3->command(opl3, opl3_reg, reg_val); } } sound/drivers/opl3/opl3_midi.c:399:3: warning: Value stored to 'vp2' is never read [deadcode.DeadStores] vp2 = &opl3->voices[voice + 3]; ^ ~~~~~~~~~~~~~~~~~~~~~~~~ I suspect that references to vp in this if block should be to vp2, but I'm unsure if that is for all references or not, hence I'm reporting this issue. Coln

2 1

[PATCH] ALSA: hda/realtek: Amp init fixup for HP ZBook 15 G6
by Bradley Scott 12 Dec '21

12 Dec '21

HP ZBook 15 G6 (SSID 103c:860f) needs the same speaker amplifier initialization as used on several other HP laptops using ALC285. This commit also adds a new "alc285-hp-amp-init" model that can be used to apply this same amp init fixup to other devices by passing "hda_model=alc285-hp-amp-init" to the snd-sof-intel-hda-common module or "model=alc285-hp-amp-init" to the snd-hda-intel module, depending on which is being used. Signed-off-by: Bradley Scott <Bradley.Scott(a)zebra.com> --- Documentation/sound/hd-audio/models.rst | 2 ++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 0ea967d34583..d25335993e55 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -326,6 +326,8 @@ usi-headset Headset support on USI machines dual-codecs Lenovo laptops with dual codecs +alc285-hp-amp-init + HP laptops which require speaker amplifier initialization (ALC285) ALC680 ====== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3599f4c85ebf..fc41f3e8ddc3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8660,6 +8660,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), + SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), @@ -9123,6 +9124,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, + {.id = ALC285_FIXUP_HP_GPIO_AMP_INIT, .name = "alc285-hp-amp-init"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1 ________________________________ - CONFIDENTIAL- This email and any files transmitted with it are confidential, and may also be legally privileged. If you are not the intended recipient, you may not review, use, copy, or distribute this message. If you receive this email in error, please notify the sender immediately by reply email and then delete this email.

2 1

Add profile for Behringer UMC204HD
by GitHub pull_request - opened 12 Dec '21

12 Dec '21

alsa-project/alsa-ucm-conf pull request #128 was opened from brndd: Adds a profile for the Behringer UMC204HD to split a 4-channel output into two outputs, and a 2-channel input into two inputs, as per the physical layout of the device. I have very little idea what I'm doing, so let me know if something looks incorrect (there's one thing that looks incorrect even to me which I shall point out in a comment below). The config does at least work, though. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7698277/… ----- ![UMC204HD_P0BK0_Front_XL](https://user-images.githubusercontent.com/5822375/145704001-5bb9b741-0d1f-4f83-8184-6335c9b53fbf.png) ![UMC204HD_P0BK0_Rear_XL](https://user-images.githubusercontent.com/5822375/145703999-918614e6-3298-4191-a7b6-7260e8eb3638.png) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/128 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/128.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: jack: Check the return value of kstrdup()
by xkernel 12 Dec '21

12 Dec '21

kstrdup() can return NULL, it is better to check the return value of it. Signed-off-by: xkernel <xkernel.wang(a)foxmail.com> --- sound/core/jack.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/jack.c b/sound/core/jack.c index 32350c6..5fd9954 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -509,6 +509,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; jack->id = kstrdup(id, GFP_KERNEL); + if (jack->id == NULL) + return -ENOMEM; /* don't creat input device for phantom jack */ if (!phantom_jack) { --

2 1

Move the call to expand variables and expressions
by GitHub pull_request - opened 12 Dec '21

12 Dec '21

alsa-project/alsa-utils pull request #127 was opened from ranj063: With the addition of the math expression evaluator in alsa-lib, we can use it to evaluate simple math expresionns for objects attribute values based on other attributes. For example to calculate the size attribute in buffer objects: size "$[$[$[$periods * $sample_size] * $channels] * $[$[$rate * $period]]] / 1000000" The size attribute value will be computed from the values for the periods, sample_size, channels, rate and period attribute values set in the buffer object. Request URL : https://github.com/alsa-project/alsa-utils/pull/127 Patch URL : https://github.com/alsa-project/alsa-utils/pull/127.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Is it a regression? 5.15.5: No sound with speaker-test. 5.12.13: it works.
by u34＠net9.ga 11 Dec '21

11 Dec '21

The following looks to me a regression in the linux sound driver. alsa-info.sh output is included at the bottom of this message. There are 2 outputs. An output with a 5.12.13 kernel. The other output is with a 5.15.5. They were obtained with the same PC. With 5.12.13, the speaker-test program works as expected. There is no sound when the speaker-test program is ran with 5.15.5. 5.15.5 has options snd-hda-intel id=PCH index=0 options snd-hda-intel id=HDMI index=1 , which 5.12.13 does not have. I have tried various combinations of the knobs of alsamixer. The dmesg section for the 5.15.5 kernal is 2 lines shorter then for the former kernel. In addition, the total number of lines in the output of the 5.15.5 kernel is greater, compared to the 5.12.13 output. I don't know those lines are of any significance. alsa-info output with 5.12.13: ============================== upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.5.0 !!################################ !!Script ran on: Fri Nov 26 05:51:28 UTC 2021 !!Linux Distribution !!------------------ Arch Linux \r (\l) Arch Linux release DISTRIB_ID=Arch DISTRIB_DESCRIPTION="Arch Linux" NAME="ArchBang Linux" ID=arch PRETTY_NAME="ArchBang Linux" !!DMI Information !!--------------- Manufacturer: Product Name: Product Version: Firmware Version: FLB8510H.86A.0131.2017.0516.1705 System SKU: To be filled by O.E.M. Board Vendor: Intel Corporation Board Name: DB85FL !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/INT3F0D:00/status 15 /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C01:00/status 15 /sys/bus/acpi/devices/PNP0C04:00/status 31 /sys/bus/acpi/devices/PNP0C0C:00/status 11 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/device:29/status 15 /sys/bus/acpi/devices/device:2a/status 15 /sys/bus/acpi/devices/device:2b/status 15 /sys/bus/acpi/devices/device:2c/status 15 /sys/bus/acpi/devices/device:2d/status 15 /sys/bus/acpi/devices/device:2e/status 15 /sys/bus/acpi/devices/device:2f/status 15 /sys/bus/acpi/devices/device:30/status 15 /sys/bus/acpi/devices/device:31/status 15 /sys/bus/acpi/devices/device:32/status 15 /sys/bus/acpi/devices/device:33/status 15 /sys/bus/acpi/devices/device:34/status 15 /sys/bus/acpi/devices/device:35/status 15 /sys/bus/acpi/devices/device:36/status 15 /sys/bus/acpi/devices/device:37/status 15 /sys/bus/acpi/devices/device:38/status 15 /sys/bus/acpi/devices/device:39/status 15 /sys/bus/acpi/devices/device:3a/status 15 /sys/bus/acpi/devices/device:3b/status 15 /sys/bus/acpi/devices/device:3c/status 15 /sys/bus/acpi/devices/device:3d/status 15 /sys/bus/acpi/devices/device:4e/status 15 !!Kernel Information !!------------------ Kernel release: 5.12.13-arch1-1 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k5.12.13-arch1-1 Library version: 1.2.5.1 Utilities version: 1.2.5.1 !!Loaded ALSA modules !!------------------- snd_hda_intel (card 0) snd_hda_intel (card 1) !!Sound Servers on this system !!---------------------------- Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7c34000 irq 29 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c30000 irq 30 !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06) Subsystem: Intel Corporation Device [8086:2052] 00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 04) DeviceName: CODEC2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Sysfs card info !!--------------- !!Card: /sys/class/sound/card0 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card0 |-- controlC0 | |-- dev | |-- device -> ../../card0 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:03.0 |-- hwC0D0 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card0 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input5 | |-- capabilities | |-- device -> ../../card0 | |-- event5 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input6 | |-- capabilities | |-- device -> ../../card0 | |-- event6 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input7 | |-- capabilities | |-- device -> ../../card0 | |-- event7 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC0D3p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D7p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D8p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!Card: /sys/class/sound/card1 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card1 |-- controlC1 | |-- dev | |-- device -> ../../card1 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:1b.0 |-- hwC1D2 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card1 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input10 | |-- capabilities | |-- device -> ../../card1 | |-- event10 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input8 | |-- capabilities | |-- device -> ../../card1 | |-- event8 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input9 | |-- capabilities | |-- device -> ../../card1 | |-- event9 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC1D0c | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D0p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D1p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D2c | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Intel Haswell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862807 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560030: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Realtek ALC662 rev1 Address: 2 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0662 Subsystem Id: 0x80862052 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Headphone+LO Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC662 rev1 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x11 0x11] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x04 0x0b Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x4017c040: [N/A] Speaker at Ext N/A Conn = Analog, Color = UNKNOWN DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00001734: IN OUT Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19030: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0e Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19140: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x0181303f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xf Pin-ctls: 0x20: IN Unsolicited: tag=03, enabled=1 Connection: 1 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02214150: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x40b4c601: [N/A] Telephony at Ext N/A Conn = RCA, Color = UNKNOWN DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x90460120: [Fixed] SPDIF Out at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=12 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x26 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 6 Nov 26 05:37 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 12 Nov 26 05:37 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 5 Nov 26 05:37 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 11 Nov 26 05:37 /dev/snd/hwC1D2 crw-rw----+ 1 root audio 116, 2 Nov 26 05:37 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 3 Nov 26 05:37 /dev/snd/pcmC0D7p crw-rw----+ 1 root audio 116, 4 Nov 26 05:37 /dev/snd/pcmC0D8p crw-rw----+ 1 root audio 116, 8 Nov 26 05:37 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 7 Nov 26 05:37 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 9 Nov 26 05:37 /dev/snd/pcmC1D1p crw-rw----+ 1 root audio 116, 10 Nov 26 05:37 /dev/snd/pcmC1D2c crw-rw----+ 1 root audio 116, 1 Nov 26 05:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Nov 26 05:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 Nov 26 05:37 . drwxr-xr-x 3 root root 320 Nov 26 05:37 .. lrwxrwxrwx 1 root root 12 Nov 26 05:37 pci-0000:00:03.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 Nov 26 05:37 pci-0000:00:1b.0 -> ../controlC1 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 1: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 2: ALC662 rev1 Alt Analog [ALC662 rev1 Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card HDMI Card sysdefault:0 'HDMI'/'HDA Intel HDMI at 0xf7c34000 irq 29' Mixer name : 'Intel Haswell HDMI' Components : 'HDA:80862807,80860101,00100000' Controls : 21 Simple ctrls : 3 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!-------Mixer controls for card PCH Card sysdefault:1 'PCH'/'HDA Intel PCH at 0xf7c30000 irq 30' Mixer name : 'Realtek ALC662 rev1' Components : 'HDA:10ec0662,80862052,00100101' Controls : 44 Simple ctrls : 21 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 44 [69%] [-20.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Headphone+LO',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] Front Right: Playback 64 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 17 [55%] [12.00dB] [on] Front Right: Capture 17 [55%] [12.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [off] Front Right: Capture 0 [0%] [-13.50dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!Alsactl output !!-------------- --startcollapse-- state.HDMI { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.20 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.21 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } state.PCH { control.1 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '4ch' item.2 '6ch' } } control.2 { iface MIXER name 'Headphone+LO Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.3 { iface MIXER name 'Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Surround Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.5 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Center Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.7 { iface MIXER name 'LFE Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.8 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.11 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.12 { iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.13 { iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Rear Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.15 { iface MIXER name 'Rear Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.17 { iface MIXER name 'Line Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.18 { iface MIXER name 'Input Source' value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.19 { iface MIXER name 'Input Source' index 1 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.20 { iface MIXER name 'Capture Volume' value.0 17 value.1 17 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 1200 dbvalue.1 1200 } } control.21 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.22 { iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 -1350 dbvalue.1 -1350 } } control.23 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.24 { iface MIXER name 'Front Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.25 { iface MIXER name 'Rear Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.26 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.27 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.28 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.30 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.31 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.32 { iface MIXER name 'IEC958 Default PCM Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.33 { iface MIXER name 'Master Playback Volume' value 44 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -2000 } } control.34 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.35 { iface CARD name 'Front Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.36 { iface CARD name 'Rear Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.37 { iface CARD name 'Line Jack' value false comment { access read type BOOLEAN count 1 } } control.38 { iface CARD name 'Line Out Jack' value true comment { access read type BOOLEAN count 1 } } control.39 { iface CARD name 'Front Headphone Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.40 { iface CARD name 'SPDIF Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.41 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.42 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.43 { iface PCM device 1 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.44 { iface PCM device 2 name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } } --endcollapse-- !!All Loaded Modules !!------------------ aesni_intel agpgart at24 bpf_preload cdc_ether cdrom cec coretemp crc32_pclmul crc32c_intel crct10dif_pclmul cryptd crypto_simd dm_mod drm drm_kms_helper e1000e fat fb_sys_fops fuse ghash_clmulni_intel i2c_algo_bit i2c_i801 i2c_smbus i915 iTCO_vendor_support iTCO_wdt intel_cstate intel_gtt intel_pmc_bxt intel_powerclamp intel_rapl_common intel_rapl_msr intel_uncore ip_tables irqbypass isofs kvm kvm_intel ledtrig_audio loop lpc_ich mac_hid mei mei_hdcp mei_me mei_wdt mii mousedev overlay rapl rfkill snd snd_hda_codec snd_hda_codec_generic snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_core snd_hda_intel snd_hwdep snd_intel_dspcfg snd_intel_sdw_acpi snd_pcm snd_timer soundcore squashfs sr_mod syscopyarea sysfillrect sysimgblt uas usb_storage usbhid usbnet vfat video x86_pkg_temp_thermal x_tables xhci_pci xhci_pci_renesas !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x05 0x58560010 0x06 0x58560020 0x07 0x18560030 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC1D2/init_pin_configs: 0x14 0x01014010 0x15 0x4017c040 0x16 0x411111f0 0x18 0x01a19030 0x19 0x02a19140 0x1a 0x0181303f 0x1b 0x02214150 0x1c 0x411111f0 0x1d 0x40b4c601 0x1e 0x90460120 /sys/class/sound/hwC1D2/driver_pin_configs: /sys/class/sound/hwC1D2/user_pin_configs: /sys/class/sound/hwC1D2/init_verbs: /sys/class/sound/hwC1D2/hints: !!ALSA/HDA dmesg !!-------------- [ 0.089835] ACPI: Added _OSI(Linux-Dell-Video) [ 0.089835] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) [ 0.089835] ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) -- [ 13.947834] iTCO_vendor_support: vendor-support=0 [ 14.140544] snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) [ 14.140729] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 14.141149] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 14.186225] iTCO_wdt iTCO_wdt.1.auto: Found a Lynx Point TCO device (Version=2, TCOBASE=0x1860) [ 14.186295] iTCO_wdt iTCO_wdt.1.auto: initialized. heartbeat=30 sec (nowayout=0) [ 14.354484] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input5 [ 14.354528] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input6 [ 14.354556] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input7 [ 14.584436] snd_hda_codec_realtek hdaudioC1D2: autoconfig for ALC662 rev1: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 14.584441] snd_hda_codec_realtek hdaudioC1D2: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 14.584443] snd_hda_codec_realtek hdaudioC1D2: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 14.584444] snd_hda_codec_realtek hdaudioC1D2: mono: mono_out=0x0 [ 14.584445] snd_hda_codec_realtek hdaudioC1D2: dig-out=0x1e/0x0 [ 14.584446] snd_hda_codec_realtek hdaudioC1D2: inputs: [ 14.584447] snd_hda_codec_realtek hdaudioC1D2: Front Mic=0x19 [ 14.584448] snd_hda_codec_realtek hdaudioC1D2: Rear Mic=0x18 [ 14.584449] snd_hda_codec_realtek hdaudioC1D2: Line=0x1a [ 14.596570] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input8 [ 14.596620] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9 [ 14.596661] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card1/input10 [ 14.936847] intel_rapl_common: Found RAPL domain package alsa-info output with 5.15.5: ============================= upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.5.0 !!################################ !!Script ran on: Fri Nov 26 23:39:13 UTC 2021 !!Linux Distribution !!------------------ Arch Linux \r (\l) NAME="Arch Linux" PRETTY_NAME="Arch Linux" ID=arch HOME_URL="https://archlinux.org/" DOCUMENTATION_URL="https://wiki.archlinux.org/" SUPPORT_URL="https://bbs.archlinux.org/" BUG_REPORT_URL="https://bugs.archlinux.org/" LOGO=archlinux-logo !!DMI Information !!--------------- Manufacturer: Product Name: Product Version: Firmware Version: FLB8510H.86A.0131.2017.0516.1705 System SKU: To be filled by O.E.M. Board Vendor: Intel Corporation Board Name: DB85FL !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/INT3F0D:00/status 15 /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C01:00/status 15 /sys/bus/acpi/devices/PNP0C04:00/status 31 /sys/bus/acpi/devices/PNP0C0C:00/status 11 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/device:29/status 15 /sys/bus/acpi/devices/device:2a/status 15 /sys/bus/acpi/devices/device:2b/status 15 /sys/bus/acpi/devices/device:2c/status 15 /sys/bus/acpi/devices/device:2d/status 15 /sys/bus/acpi/devices/device:2e/status 15 /sys/bus/acpi/devices/device:2f/status 15 /sys/bus/acpi/devices/device:30/status 15 /sys/bus/acpi/devices/device:31/status 15 /sys/bus/acpi/devices/device:32/status 15 /sys/bus/acpi/devices/device:33/status 15 /sys/bus/acpi/devices/device:34/status 15 /sys/bus/acpi/devices/device:35/status 15 /sys/bus/acpi/devices/device:36/status 15 /sys/bus/acpi/devices/device:37/status 15 /sys/bus/acpi/devices/device:38/status 15 /sys/bus/acpi/devices/device:39/status 15 /sys/bus/acpi/devices/device:3a/status 15 /sys/bus/acpi/devices/device:3b/status 15 /sys/bus/acpi/devices/device:3c/status 15 /sys/bus/acpi/devices/device:3d/status 15 /sys/bus/acpi/devices/device:4e/status 15 !!Kernel Information !!------------------ Kernel release: 5.15.5-arch1-1 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k5.15.5-arch1-1 Library version: 1.2.5.1 Utilities version: 1.2.5.1 !!Loaded ALSA modules !!------------------- snd_hda_intel (card 0) snd_hda_intel (card 1) !!Sound Servers on this system !!---------------------------- Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c30000 irq 30 1 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7c34000 irq 29 !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06) Subsystem: Intel Corporation Device [8086:2052] 00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 04) DeviceName: CODEC2 !!Modprobe options (Sound related) !!-------------------------------- snd_hda_intel: id=PCH index=0 snd_hda_intel: id=HDMI index=1 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : HDMI,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : HDMI,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Sysfs card info !!--------------- !!Card: /sys/class/sound/card0 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card0 |-- controlC0 | |-- dev | |-- device -> ../../card0 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:1b.0 |-- hwC0D2 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card0 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input10 | |-- capabilities | |-- device -> ../../card0 | |-- event10 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input11 | |-- capabilities | |-- device -> ../../card0 | |-- event11 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input9 | |-- capabilities | |-- device -> ../../card0 | |-- event9 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC0D0c | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D0p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D1p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D2c | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!Card: /sys/class/sound/card1 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card1 |-- controlC1 | |-- dev | |-- device -> ../../card1 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:03.0 |-- hwC1D0 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card1 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input6 | |-- capabilities | |-- device -> ../../card1 | |-- event6 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input7 | |-- capabilities | |-- device -> ../../card1 | |-- event7 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input8 | |-- capabilities | |-- device -> ../../card1 | |-- event8 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC1D3p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D7p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D8p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC662 rev1 Address: 2 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0662 Subsystem Id: 0x80862052 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Headphone+LO Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=5, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x1c 0x1c] Converter: stream=5, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x1c 0x1c] Converter: stream=5, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC662 rev1 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x04 0x0b Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x4017c040: [N/A] Speaker at Ext N/A Conn = Analog, Color = UNKNOWN DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00001734: IN OUT Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19030: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0e Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19140: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x0181303f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xf Pin-ctls: 0x20: IN Unsolicited: tag=03, enabled=1 Connection: 1 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02214150: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x40b4c601: [N/A] Telephony at Ext N/A Conn = RCA, Color = UNKNOWN DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x90460120: [Fixed] SPDIF Out at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=12 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x26 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Intel Haswell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862807 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560030: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 12 Nov 26 23:37 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 6 Nov 26 23:37 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 11 Nov 26 23:37 /dev/snd/hwC0D2 crw-rw----+ 1 root audio 116, 5 Nov 26 23:37 /dev/snd/hwC1D0 crw-rw----+ 1 root audio 116, 8 Nov 26 23:37 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 7 Nov 26 23:38 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 9 Nov 26 23:37 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 10 Nov 26 23:37 /dev/snd/pcmC0D2c crw-rw----+ 1 root audio 116, 2 Nov 26 23:37 /dev/snd/pcmC1D3p crw-rw----+ 1 root audio 116, 3 Nov 26 23:37 /dev/snd/pcmC1D7p crw-rw----+ 1 root audio 116, 4 Nov 26 23:37 /dev/snd/pcmC1D8p crw-rw----+ 1 root audio 116, 1 Nov 26 23:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Nov 26 23:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 Nov 26 23:37 . drwxr-xr-x 3 root root 320 Nov 26 23:37 .. lrwxrwxrwx 1 root root 12 Nov 26 23:37 pci-0000:00:03.0 -> ../controlC1 lrwxrwxrwx 1 root root 12 Nov 26 23:37 pci-0000:00:1b.0 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC662 rev1 Alt Analog [ALC662 rev1 Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card PCH Card sysdefault:0 'PCH'/'HDA Intel PCH at 0xf7c30000 irq 30' Mixer name : 'Realtek ALC662 rev1' Components : 'HDA:10ec0662,80862052,00100101' Controls : 45 Simple ctrls : 22 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 46 [72%] [-18.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Headphone+LO',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 0 [0%] [-64.00dB] Front Right: Playback 0 [0%] [-64.00dB] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 173 [68%] [-16.40dB] Front Right: Playback 173 [68%] [-16.40dB] Simple mixer control 'Front',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 46 [72%] [-18.00dB] [off] Front Right: Playback 46 [72%] [-18.00dB] [off] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 46 [72%] [-18.00dB] [off] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 46 [72%] [-18.00dB] [off] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 20 [65%] [-4.50dB] [on] Front Right: Playback 20 [65%] [-4.50dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [on] Front Right: Capture 0 [0%] [-13.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [off] Front Right: Capture 0 [0%] [-13.50dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!-------Mixer controls for card HDMI Card sysdefault:1 'HDMI'/'HDA Intel HDMI at 0xf7c34000 irq 29' Mixer name : 'Intel Haswell HDMI' Components : 'HDA:80862807,80860101,00100000' Controls : 21 Simple ctrls : 3 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '4ch' item.2 '6ch' } } control.2 { iface MIXER name 'Headphone+LO Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -6400 dbvalue.1 -6400 } } control.3 { iface MIXER name 'Front Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Surround Playback Volume' value.0 46 value.1 46 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 dbvalue.1 -1800 } } control.5 { iface MIXER name 'Surround Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Center Playback Volume' value 46 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 } } control.7 { iface MIXER name 'LFE Playback Volume' value 46 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 } } control.8 { iface MIXER name 'Center Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'LFE Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.11 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.12 { iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.13 { iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Rear Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.15 { iface MIXER name 'Rear Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Line Playback Volume' value.0 20 value.1 20 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -450 dbvalue.1 -450 } } control.17 { iface MIXER name 'Line Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.18 { iface MIXER name 'Input Source' value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.19 { iface MIXER name 'Input Source' index 1 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.20 { iface MIXER name 'Capture Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 -1350 dbvalue.1 -1350 } } control.21 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.22 { iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 -1350 dbvalue.1 -1350 } } control.23 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.24 { iface MIXER name 'Front Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.25 { iface MIXER name 'Rear Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.26 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.27 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.28 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.30 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.31 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.32 { iface MIXER name 'IEC958 Default PCM Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.33 { iface MIXER name 'Master Playback Volume' value 46 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 } } control.34 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.35 { iface CARD name 'Front Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.36 { iface CARD name 'Rear Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.37 { iface CARD name 'Line Jack' value false comment { access read type BOOLEAN count 1 } } control.38 { iface CARD name 'Line Out Jack' value true comment { access read type BOOLEAN count 1 } } control.39 { iface CARD name 'Front Headphone Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.40 { iface CARD name 'SPDIF Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.41 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.42 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.43 { iface PCM device 1 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.44 { iface PCM device 2 name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.45 { iface MIXER name 'PCM Playback Volume' value.0 173 value.1 173 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -1640 dbvalue.1 -1640 } } } state.HDMI { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value false comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value false comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.20 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.21 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } --endcollapse-- !!All Loaded Modules !!------------------ aesni_intel at24 cdc_ether cdrom coretemp crc16 crc32_pclmul crc32c_generic crc32c_intel crct10dif_pclmul cryptd crypto_simd dm_mod e1000e ext4 fat fuse ghash_clmulni_intel i2c_i801 i2c_smbus i915 iTCO_vendor_support iTCO_wdt intel_cstate intel_gtt intel_pmc_bxt intel_powerclamp intel_rapl_common intel_rapl_msr intel_spi intel_spi_platform intel_uncore ip_tables irqbypass jbd2 kvm kvm_intel ledtrig_audio libcrc32c lpc_ich mac_hid mbcache mei mei_hdcp mei_me mei_wdt mii mousedev mtd nct6683 nf_conntrack nf_defrag_ipv4 nf_defrag_ipv6 nf_reject_ipv4 nf_reject_ipv6 nf_tables nfnetlink nft_counter nft_ct nft_limit nft_reject nft_reject_inet nls_iso8859_1 pcspkr rapl sg snd snd_hda_codec snd_hda_codec_generic snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_core snd_hda_intel snd_hwdep snd_intel_dspcfg snd_intel_sdw_acpi snd_pcm snd_timer soundcore spi_nor sr_mod ttm uas usb_storage usbhid usbnet vfat video x86_pkg_temp_thermal x_tables xhci_pci xhci_pci_renesas !!Sysfs Files !!----------- /sys/class/sound/hwC0D2/init_pin_configs: 0x14 0x01014010 0x15 0x4017c040 0x16 0x411111f0 0x18 0x01a19030 0x19 0x02a19140 0x1a 0x0181303f 0x1b 0x02214150 0x1c 0x411111f0 0x1d 0x40b4c601 0x1e 0x90460120 /sys/class/sound/hwC0D2/driver_pin_configs: /sys/class/sound/hwC0D2/user_pin_configs: /sys/class/sound/hwC0D2/init_verbs: /sys/class/sound/hwC0D2/hints: /sys/class/sound/hwC1D0/init_pin_configs: 0x05 0x58560010 0x06 0x58560020 0x07 0x18560030 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: !!ALSA/HDA dmesg !!-------------- [ 0.099082] ACPI: Added _OSI(Linux-Dell-Video) [ 0.099082] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) [ 0.099082] ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) -- [ 12.324700] AES CTR mode by8 optimization enabled [ 12.404051] snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) [ 12.404225] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 12.404685] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 12.535527] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card1/input6 [ 12.535599] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card1/input7 [ 12.535660] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card1/input8 [ 12.877211] snd_hda_codec_realtek hdaudioC0D2: autoconfig for ALC662 rev1: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 12.877227] snd_hda_codec_realtek hdaudioC0D2: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 12.877233] snd_hda_codec_realtek hdaudioC0D2: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 12.877239] snd_hda_codec_realtek hdaudioC0D2: mono: mono_out=0x0 [ 12.877243] snd_hda_codec_realtek hdaudioC0D2: dig-out=0x1e/0x0 [ 12.877247] snd_hda_codec_realtek hdaudioC0D2: inputs: [ 12.877251] snd_hda_codec_realtek hdaudioC0D2: Front Mic=0x19 [ 12.877256] snd_hda_codec_realtek hdaudioC0D2: Rear Mic=0x18 [ 12.877260] snd_hda_codec_realtek hdaudioC0D2: Line=0x1a [ 12.893002] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 12.893066] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 12.893118] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 13.779298] intel_rapl_common: Found RAPL domain package -- u34

4 5

Audio output not working at all on Fedora all in alsa 1.2.6
by GitHub issues - opened 11 Dec '21

11 Dec '21

alsa-project/alsa-lib issue #202 was opened from sstraightea: After upgrading to the newest version of alsa GNOME shows me I do not have any audio input device. The microphone seems to still be working for me. Downgrading solves the problem. I have been able to reproduce this problem on two different machines. Just install the newest Fedora and do a normal update. My system: Fedora 35 GNOME 41.2 >From dnf info alsa-lib: ``` Name : alsa-lib Version : 1.2.6 Release : 1.fc35 Architecture : x86_64 Size : 1.4 M Source : alsa-lib-1.2.6-1.fc35.src.rpm Repository : @System >From repo : updates ``` If you need any log files, configs, output of some command I am happy to provide that, but I am not sure what information you need. Issue URL : https://github.com/alsa-project/alsa-lib/issues/202 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Permissions of alsatplg -o output are hardcoded to S_IRUSR | S_IWUSR (600)
by GitHub issues - opened 11 Dec '21

11 Dec '21

alsa-project/alsa-utils issue #126 was opened from marc-hb: >From https://github.com/alsa-project/alsa-utils/blob/c8c348e28a258f17e3/topology… ```C static int save(const char *output_file, void *buf, size_t size) { fd = open(fname, O_RDWR | O_CREAT | O_TRUNC, S_IRUSR | S_IWUSR); ``` When using alsatplg in a build system (e.g.: https://github.com/thesofproject/sof/blob/3e1c160e5053eae/tools/topology/to…), all other artefacts follow the process's umask **except** alsatplg output. This causes subtle and time-consuming errors when trying to share artefacts across different processes. git blame finds: commit cbabe7a3f0cc84ecd352d4cbf85148946fa6c0d5 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Thu Dec 19 15:36:05 2019 +0100 alsatplg: rewrite to use the new libatopology functions Before cbabe7a3f0, opening the output file was delegated to library function `snd_tplg_build_file()`. Not sure what permissions were used then. Workaround: ```shell alsatplg -c intput.conf -o - > output.tplg ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/126 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v6] dt-bindings: dsp: mediatek: Add mt8195 DSP binding support
by YC Hung 10 Dec '21

10 Dec '21

From: "yc.hung" <yc.hung(a)mediatek.com> This describes the mt8195 DSP device tree node. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- Changes since v5: Remove sub node "sound" and "mediatek,platform", "mediatek,dptx-codec", "mediatek,hdmi-codec" properties. Let dsp node as an individual node and sound card has phandle of this dsp node per discussion in https://github.com/thesofproject/linux/pull/3239 Changes since v4: - remove patch 1 with clocks as it will be reviewed and merged to SOF tree. https://github.com/thesofproject/linux/pull/3237 - refine DT file binding files with Rob's comments. --- .../bindings/dsp/mtk,mt8195-dsp.yaml | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml index f8bacd50b018..33db11ee2336 100644 --- a/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml @@ -70,22 +70,6 @@ properties: - description: A phandle to a reserved memory region is used for dma buffer between host and DSP. - description: A phandle to a reserved memory region is used for DSP system memory. - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - mediatek,platform: - $ref: "/schemas/types.yaml#/definitions/phandle" - description: The phandle of MT8195 ASoC platform. - - mediatek,dptx-codec: - $ref: "/schemas/types.yaml#/definitions/phandle" - description: The phandle of MT8195 Display Port Tx codec node. - - mediatek,hdmi-codec: - $ref: "/schemas/types.yaml#/definitions/phandle" - description: The phandle of MT8195 HDMI codec node. - required: - compatible - reg @@ -98,7 +82,6 @@ required: - power-domains - mbox-names - mboxes - - mediatek,platform additionalProperties: false @@ -130,6 +113,4 @@ examples: power-domains = <&spm 6>; //MT8195_POWER_DOMAIN_ADSP mbox-names = "mbox0", "mbox1"; mboxes = <&adsp_mailbox0>, <&adsp_mailbox1>; - mediatek,dptx-codec = <&dp_tx>; - mediatek,platform = <&afe>; }; -- 2.18.0

2 1

[PATCH 0/2] This patches provide ASoc codec support for ES8156
by Shumin Chen 10 Dec '21

10 Dec '21

Introduce new ASoc codec ES8156 support Shumin Chen (2): ASoC: add ES8156 codec driver ASoC: convert Everest ES8156 binding to yaml .../bindings/sound/everest,es8156.yaml | 49 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8156.c | 614 ++++++++++++++++++ sound/soc/codecs/es8156.h | 76 +++ 5 files changed, 746 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es8156.yaml create mode 100644 sound/soc/codecs/es8156.c create mode 100644 sound/soc/codecs/es8156.h -- 2.25.1

4 7

ALSA 1.2.6.2 release
by Jaroslav Kysela 10 Dec '21

10 Dec '21

Hello all, new ALSA userspace packages were released. This is second bugfix release. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-ucm-conf Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.6.1_v1.2.6.2 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

Add UCM support for SDM845 based Lenovo yoga c630
by GitHub pull_request - edited 10 Dec '21

10 Dec '21

alsa-project/alsa-ucm-conf pull request #88 was edited from Srinivas-Kandagatla: One of the Lenovo YOGA C360 variant is based off Qualcomm SDM845 SoC. This PR adds support to On Board Speakers and Headset. This PR also has a small change to the codec Headphone sequences to be inline with other Sequences. thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/88 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/88.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] mfd: wcd934x: fix typo and a duplicate register define
by Srinivas Kandagatla 10 Dec '21

10 Dec '21

Looks like we ended up with a typo and a duplicate in WCD register defines. Fix these. Fixes: 6ac7e4d7ad70 ("mfd: wcd934x: Add support to wcd9340/wcd9341 codec") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- include/linux/mfd/wcd934x/registers.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/linux/mfd/wcd934x/registers.h b/include/linux/mfd/wcd934x/registers.h index 76a943c83c63..bcec70a66fff 100644 --- a/include/linux/mfd/wcd934x/registers.h +++ b/include/linux/mfd/wcd934x/registers.h @@ -34,7 +34,6 @@ #define WCD934X_DMIC_RATE_MASK GENMASK(3, 1) #define WCD934X_CPE_SS_DMIC2_CTL 0x021a #define WCD934X_CPE_SS_DMIC_CFG 0x021b -#define WCD934X_CPE_SS_DMIC_CFG 0x021b #define WCD934X_CPE_SS_CPAR_CFG 0x021c #define WCD934X_INTR_PIN1_MASK0 0x0409 #define WCD934X_INTR_PIN1_STATUS0 0x0411 @@ -375,7 +374,7 @@ #define WCD934X_CDC_RX2_RX_PATH_SEC3 0x0b74 #define WCD934X_CDC_RX2_RX_PATH_DSMDEM_CTL 0x0b7b #define WCD934X_CDC_RX3_RX_PATH_CTL 0x0b7d -#define WCD934X_CDC_RX3_RX_PATH_CFG0 0x0b6e +#define WCD934X_CDC_RX3_RX_PATH_CFG0 0x0b7e #define WCD934X_CDC_RX3_RX_PATH_CFG2 0x0b80 #define WCD934X_CDC_RX3_RX_VOL_CTL 0x0b81 #define WCD934X_CDC_RX3_RX_PATH_MIX_CTL 0x0b82 -- 2.21.0

1 0

Amlogic DPCM audio card(s) speaker placement issue
by Christian Hewitt 10 Dec '21

10 Dec '21

Hello alsa-dev / linux-amlogic, I’d like to describe a problem with speaker placement seen with the Amlogic (meson) AIU and AXG audio drivers. Playing media in Kodi I’d noticed that placement is consistently wrong; meaning the incorrect placement remains the same over different media and Amlogic hardwares. This was traced to a failure to get the channel map via snd_pcm_get_chmap() and further analysis showed this failed because the PCM controls are exposed on a different device than the PCM stream. e.g. on LibreTech LePotato with DPCM routing configured for HDMI output, the hw PCM is on the single frontend device 0 but the PCM controls are on device 2: LePotato:~ # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: LIBRETECHCC [LIBRETECH-CC], device 0: fe.dai-link-0 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 LePotato:~ # amixer controls numid=19,iface=MIXER,name='PCM Playback Volume' numid=18,iface=MIXER,name='ACODEC Left DAC Sel' numid=7,iface=MIXER,name='ACODEC Mute Ramp Switch' numid=2,iface=MIXER,name='ACODEC Playback Channel Mode' numid=3,iface=MIXER,name='ACODEC Playback Switch' numid=4,iface=MIXER,name='ACODEC Playback Volume' numid=5,iface=MIXER,name='ACODEC Ramp Rate' numid=17,iface=MIXER,name='ACODEC Right DAC Sel' numid=8,iface=MIXER,name='ACODEC Unmute Ramp Switch' numid=6,iface=MIXER,name='ACODEC Volume Ramp Switch' numid=1,iface=MIXER,name='AIU ACODEC I2S Lane Select' numid=16,iface=MIXER,name='AIU ACODEC OUT EN Switch' numid=15,iface=MIXER,name='AIU ACODEC SRC' numid=14,iface=MIXER,name='AIU HDMI CTRL SRC' numid=13,iface=MIXER,name='AIU SPDIF SRC SEL' numid=12,iface=PCM,name='ELD',device=2 numid=11,iface=PCM,name='IEC958 Playback Default',device=2 numid=10,iface=PCM,name='IEC958 Playback Mask',device=2 numid=9,iface=PCM,name='Playback Channel Map',device=2 https://github.com/torvalds/linux/blob/master/arch/arm64/boot/dts/amlogic/m… or on a Wetek Play 2: WP2:~ # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: WETEKPLAY2 [WETEK-PLAY2], device 0: fe.dai-link-0 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: WETEKPLAY2 [WETEK-PLAY2], device 1: fe.dai-link-1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 WP2:~ # amixer controls numid=7,iface=MIXER,name='PCM Playback Volume' numid=6,iface=MIXER,name='AIU HDMI CTRL SRC' numid=5,iface=MIXER,name='AIU SPDIF SRC SEL' numid=4,iface=PCM,name='ELD',device=4 numid=3,iface=PCM,name='IEC958 Playback Default',device=4 numid=2,iface=PCM,name='IEC958 Playback Mask',device=4 numid=1,iface=PCM,name='Playback Channel Map',device=4 https://github.com/torvalds/linux/blob/master/arch/arm64/boot/dts/amlogic/m… It looks like the PCM controls of the backend hdmi-codec are not routed through the DPCM driver the same way as PCM streams. We discussed that issue with Jerome, the driver author, but it’s not clear which bits we are missing. i.e. if it's something ASoC DPCM should be handling or if some further plumbing in the card driver or userspace is needed? Any hints into the right direction would be highly appreciated! Christian

1 0

PA reports possible bug in 'snd_usb_audio'
by GitHub issues - opened 10 Dec '21

10 Dec '21

alsa-project/alsa-lib issue #201 was opened from riveravaldez: Hi, I'm having some hangs with an until-yesterday perfectly-working setup. Only way to fix it seems to be restarting JACK. Looking at system messages I found: ``` $ sudo journalctl -b -exp3 - Journal begins at Mon 2020-09-07 22:43:48 -03, ends at Thu 2021-12-09 17:38:30 -03. -- dic 02 20:05:13 debian pulseaudio[73378]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Burr-Brown_from_TI_USB_Audio_CODEC-00" card_name="> dic 03 14:31:49 debian pulseaudio[92221]: Failed to find a working profile. dic 03 14:31:49 debian pulseaudio[92221]: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-Evolution_Electronics_Ltd._USB_Keystation_61es-00"> dic 03 16:54:00 debian kernel: usb 1-1.2: urb status -32 dic 03 16:54:00 debian kernel: usb 1-1.2: urb status -32 (...) dic 03 16:54:02 debian kernel: usb 1-1.2: urb status -32 dic 03 16:54:02 debian kernel: usb 1-1.2: urb status -32 dic 03 16:56:37 debian pulseaudio[92221]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Burr-Brown_from_TI_USB_Audio_CODEC-00" card_name="> dic 04 13:04:55 debian pulseaudio[95017]: ALSA woke us up to write new data to the device, but there was actually nothing to write. dic 04 13:04:55 debian pulseaudio[95017]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. dic 04 13:04:55 debian pulseaudio[95017]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. dic 08 10:47:20 debian pulseaudio[174863]: Failed to find a working profile. dic 08 10:47:20 debian pulseaudio[174863]: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-Evolution_Electronics_Ltd._USB_Keystation_61es-00> dic 08 18:16:06 debian kernel: usb 1-1.2: urb status -32 dic 08 18:16:07 debian kernel: usb 1-1.2: urb status -32 (...) dic 08 18:16:08 debian kernel: usb 1-1.2: urb status -32 dic 08 18:16:08 debian kernel: usb 1-1.2: urb status -32 dic 08 18:17:50 debian pulseaudio[174863]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Burr-Brown_from_TI_USB_Audio_CODEC-00" card_name=> dic 09 13:10:40 debian pulseaudio[202715]: Failed to find a working profile. dic 09 13:10:40 debian pulseaudio[202715]: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-Evolution_Electronics_Ltd._USB_Keystation_61es-00> dic 09 13:59:46 debian pulseaudio[202715]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Burr-Brown_from_TI_USB_Audio_CODEC-00" card_name=> dic 09 14:13:31 debian pulseaudio[204356]: Failed to find a working profile. dic 09 14:13:31 debian pulseaudio[204356]: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-Evolution_Electronics_Ltd._USB_Keystation_61es-00> dic 09 14:13:52 debian pulseaudio[204395]: Failed to find a working profile. dic 09 14:13:52 debian pulseaudio[204395]: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-Evolution_Electronics_Ltd._USB_Keystation_61es-00> dic 09 14:31:20 debian pulseaudio[204395]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Burr-Brown_from_TI_USB_Audio_CODEC-00" card_name=> dic 09 14:38:26 debian pulseaudio[205228]: Failed to find a working profile. dic 09 14:38:26 debian pulseaudio[205228]: Failed to load module "module-alsa-card" (argument: "device_id="2" name="usb-Evolution_Electronics_Ltd._USB_Keystation_61es-00> dic 09 15:05:37 debian pulseaudio[205228]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Burr-Brown_from_TI_USB_Audio_CODEC-00" card_name=> dic 09 17:25:21 debian pulseaudio[205228]: jack_client_open() failed. dic 09 17:25:21 debian pulseaudio[205228]: Failed to load module "module-jack-sink" (argument: ""): initialization failed. ``` This is a ThinkPad X220T with updated Debian 11 (Stable) with the internal audio-interface turned-off through PA and an external Behringer U-PHORIA UMC22 handling all the audio. ``` $ uname -a Linux debian 5.10.0-9-amd64 #1 SMP Debian 5.10.70-1 (2021-09-30) x86_64 GNU/Linux $ lspci 00:00.0 Host bridge: Intel Corporation 2nd Generation Core Processor Family DRAM Controller (rev 09) 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 00:16.0 Communication controller: Intel Corporation 6 Series/C200 Series Chipset Family MEI Controller #1 (rev 04) 00:16.3 Serial controller: Intel Corporation 6 Series/C200 Series Chipset Family KT Controller (rev 04) 00:19.0 Ethernet controller: Intel Corporation 82579LM Gigabit Network Connection (Lewisville) (rev 04) 00:1a.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b4) 00:1c.1 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 2 (rev b4) 00:1c.3 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 4 (rev b4) 00:1c.4 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 5 (rev b4) 00:1d.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation QM67 Express Chipset LPC Controller (rev 04) 00:1f.2 SATA controller: Intel Corporation 6 Series/C200 Series Chipset Family 6 port Mobile SATA AHCI Controller (rev 04) 00:1f.3 SMBus: Intel Corporation 6 Series/C200 Series Chipset Family SMBus Controller (rev 04) 03:00.0 Network controller: Intel Corporation Centrino Advanced-N 6205 [Taylor Peak] (rev 34) 0d:00.0 System peripheral: Ricoh Co Ltd PCIe SDXC/MMC Host Controller (rev 07) $ lsusb Bus 002 Device 004: ID 056a:00e6 Wacom Co., Ltd TPCE6 Bus 002 Device 024: ID 0000:3825 USB OPTICAL MOUSE Bus 002 Device 028: ID 08bb:2902 Texas Instruments PCM2902 Audio Codec Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 003: ID 04f2:b217 Chicony Electronics Co., Ltd Lenovo Integrated Camera (0.3MP) Bus 001 Device 034: ID 0a4d:0091 Evolution Electronics, Ltd Keystation 61es Driver Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub ``` Sorry if this is useless or out of place. I'm reporting just in case (considering the PA message about a possible ALSA driver bug). Thanks a lot in advance. Any recommendation would be most welcome. Kind regards. Issue URL : https://github.com/alsa-project/alsa-lib/issues/201 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 0/3] kselftest: alsa: Add basic mixer selftest
by Mark Brown 10 Dec '21

10 Dec '21

This series adds a basic selftest for the mixer interface, as discussed in the cover letter for the main patch there's plenty of additional coverage that could be added but this is a good starting point. v3: - Pull in incremental updates adding a fixed library configuration from Jaroslav and support for volatile controls from Takashi Sakamoto. - Stylistic updates suggested by Shuah. v2: - Use pkg-config to get CFLAGS and LDLIBS for alsa-lib. Jaroslav Kysela (1): kselftest: alsa: Use private alsa-lib configuration in mixer test Mark Brown (1): kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest Takashi Sakamoto (1): kselftest: alsa: optimization for SNDRV_CTL_ELEM_ACCESS_VOLATILE MAINTAINERS | 8 + tools/testing/selftests/Makefile | 3 +- tools/testing/selftests/alsa/.gitignore | 1 + tools/testing/selftests/alsa/Makefile | 9 + tools/testing/selftests/alsa/mixer-test.c | 663 ++++++++++++++++++++++ 5 files changed, 683 insertions(+), 1 deletion(-) create mode 100644 tools/testing/selftests/alsa/.gitignore create mode 100644 tools/testing/selftests/alsa/Makefile create mode 100644 tools/testing/selftests/alsa/mixer-test.c base-commit: fa55b7dcdc43c1aa1ba12bca9d2dd4318c2a0dbf -- 2.30.2

3 8

microphone disappeared on alsa-ucm-conf 1.2.6.1
by GitHub issues - edited 10 Dec '21

10 Dec '21

alsa-project/alsa-lib issue #200 was edited from galibozek: after upgrade alsa-ucm-conf 1.2.5.1-1 to 1.2.6.1-1 (arch linux) my microphone disappeared on lenovo t14 gen2i Similar to: https://bugs.archlinux.org/task/71180 ``` 00:1f.3 Audio device: Intel Corporation Tiger Lake-LP Smart Sound Technology Audio Controller (rev 20) (prog-if 80) Subsystem: Lenovo Device 22c9 Flags: bus master, fast devsel, latency 64, IRQ 187, IOMMU group 12 Memory at 603dbc0000 (64-bit, non-prefetchable) [size=16K] Memory at 603da00000 (64-bit, non-prefetchable) [size=1M] Capabilities: <access denied> Kernel driver in use: sof-audio-pci-intel-tgl Kernel modules: snd_hda_intel, snd_sof_pci_intel_tgl ``` ``` kernel: input: sof-hda-dsp Mic as /devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0/input9 kernel: input: sof-hda-dsp Headphone as /devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0/input10 kernel: input: sof-hda-dsp HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0/input11 kernel: input: sof-hda-dsp HDMI/DP,pcm=4 as /devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0/input12 kernel: input: sof-hda-dsp HDMI/DP,pcm=5 as /devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0/input13 systemd-udevd[513]: controlC0: Process '/usr/bin/alsactl restore 0' failed with exit code 99. alsactl[1136]: alsa-lib main.c:1412:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 alsactl[1136]: alsa-lib main.c:1412:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 alsactl[1136]: Found hardware: "sof-hda-dsp" "Realtek ALC257" "HDA:80862812,80860101,00100000 HDA:10ec0257,17aa22c9,00100001 cfg-dmics:2" "" "" alsactl[1136]: Hardware is initialized using a generic method audit[1]: SERVICE_START pid=1 uid=0 auid=4294967295 ses=4294967295 msg='unit=alsa-restore comm="systemd" exe="/usr/lib/systemd/systemd" hostname=? addr=? terminal=? res=success' plasmashell[1688]: org.kde.plasma.pulseaudio: No object for name "alsa_output.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.stereo-fallback.monitor" kernel: snd_hda_codec_realtek ehdaudio0D0: didn't find PCM for DAI Digital Codec DAI kernel: snd_hda_codec_realtek ehdaudio0D0: ASoC: error at snd_soc_dai_startup on Digital Codec DAI: -22 kernel: snd_hda_codec_realtek ehdaudio0D0: didn't find PCM for DAI Digital Codec DAI kernel: snd_hda_codec_realtek ehdaudio0D0: ASoC: error at snd_soc_dai_startup on Digital Codec DAI: -22 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/200 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sof-hda-dsp: fix path to Hdmi.conf
by GitHub pull_request - opened 10 Dec '21

10 Dec '21

alsa-project/alsa-ucm-conf pull request #127 was opened from sustmi: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/126 . Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/127 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/127.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

microphone disappeared on alsa-ucm-conf 1.2.6.1
by GitHub issues - edited 10 Dec '21

10 Dec '21

alsa-project/alsa-lib issue #200 was edited from galibozek: after upgrade alsa-ucm-conf 1.2.5.1-1 to 1.2.6.1-1 (arch linux) my microphone disappeared on lenovo t14 gen2i Similar to: https://bugs.archlinux.org/task/71180 ``` 00:1f.3 Audio device: Intel Corporation Tiger Lake-LP Smart Sound Technology Audio Controller (rev 20) (prog-if 80) Subsystem: Lenovo Device 22c9 Flags: bus master, fast devsel, latency 64, IRQ 187, IOMMU group 12 Memory at 603dbc0000 (64-bit, non-prefetchable) [size=16K] Memory at 603da00000 (64-bit, non-prefetchable) [size=1M] Capabilities: <access denied> Kernel driver in use: sof-audio-pci-intel-tgl Kernel modules: snd_hda_intel, snd_sof_pci_intel_tgl ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/200 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

microphone disappeared on alsa-ucm-conf 1.2.6.1
by GitHub issues - edited 10 Dec '21

10 Dec '21

alsa-project/alsa-lib issue #200 was edited from galibozek: after upgrade alsa-ucm-conf 1.2.5.1-1 to 1.2.6.1-1 (arch linux) my microphone disappeared on lenovo t14 gen2i Similar to: https://bugs.archlinux.org/task/71180 lspci: 00:1f.3 Audio device: Intel Corporation Tiger Lake-LP Smart Sound Technology Audio Controller (rev 20) Issue URL : https://github.com/alsa-project/alsa-lib/issues/200 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

microphone disappeared on alsa-ucm-conf 1.2.6.1
by GitHub issues - edited 10 Dec '21

10 Dec '21

alsa-project/alsa-lib issue #200 was edited from galibozek: after upgrade alsa-ucm-conf 1.2.5.1-1 to 1.2.6.1-1 (arch linux) my microphone disappeared on lenovo t14 gen2i Similar to: https://bugs.archlinux.org/task/71180 Issue URL : https://github.com/alsa-project/alsa-lib/issues/200 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

microphone disappeared on alsa-ucm-conf 1.2.6.1
by GitHub issues - edited 09 Dec '21

09 Dec '21

alsa-project/alsa-lib issue #200 was edited from galibozek: after upgrade testing/alsa-ucm-conf 1.2.5.1-1 to 1.2.6.1-1 (arch linux) my microphone disappeared on lenovo t14 gen2i Similar to: https://bugs.archlinux.org/task/71180 Issue URL : https://github.com/alsa-project/alsa-lib/issues/200 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

microphone disappeared on alsa-ucm-conf 1.2.6.1
by GitHub issues - edited 09 Dec '21

09 Dec '21

alsa-project/alsa-lib issue #200 was edited from galibozek: after upgrade testing/alsa-ucm-conf 1.2.5.1-1 to 1.2.6.1-1 (arch linux) my microphone disappeared on lenovo t14 gen2i Issue URL : https://github.com/alsa-project/alsa-lib/issues/200 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

microphone disappeared on alsa-ucm-conf 1.2.6.1
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-lib issue #200 was opened from galibozek: after upgrade testing/alsa-ucm-conf 1.2.5.1-1 to 1.2.6.1-1 on arch linux my microphone disappeared on lenovo t14 gen2i Issue URL : https://github.com/alsa-project/alsa-lib/issues/200 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: SOF: Remove pm_runtime_put_autosuspend() for SOF OF device
by Daniel Baluta 09 Dec '21

09 Dec '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> In SOF OF device, pm_runtime_put_autosuspend() is not matching any pm_runtime_get_sync(). This is imbalanced for PM runtime. Also, for consistency we call pm_runtime_mark_last_busy() before enabling PM runtime. 1. Remove pm_runtime_put_autosuspend() in probe_complete 2. Reorder PM runtime calls int probe_complete Signed-off-by: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/sof-of-dev.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 412cbb824b84..b0089698eecb 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -74,11 +74,9 @@ static void sof_of_probe_complete(struct device *dev) /* allow runtime_pm */ pm_runtime_set_autosuspend_delay(dev, SND_SOF_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); pm_runtime_set_active(dev); pm_runtime_enable(dev); - - pm_runtime_mark_last_busy(dev); - pm_runtime_put_autosuspend(dev); } static int sof_of_probe(struct platform_device *pdev) -- 2.27.0

2 1

ALSA 1.2.6.1 release
by Jaroslav Kysela 09 Dec '21

09 Dec '21

Hello all, new ALSA userspace packages were released. This is a quick bugfix release. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-ucm-conf Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.6_v1.2.6.1 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[PATCH v2] ASoC: amd: Convert to new style DAI format definitions
by Mark Brown 09 Dec '21

09 Dec '21

Convert the AMD machine drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 20 ++++++++++---------- sound/soc/amd/acp-rt5645.c | 4 ++-- sound/soc/amd/acp3x-rt5682-max9836.c | 8 ++++---- 3 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b2065f3fe42c..3bf86c2424ae 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -522,7 +522,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_da7219_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -533,7 +533,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_da7219_cap_ops, @@ -543,7 +543,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_max_play_ops, @@ -554,7 +554,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic0_cap_ops, @@ -565,7 +565,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic1_cap_ops, @@ -578,7 +578,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_rt5682_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -589,7 +589,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, @@ -599,7 +599,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, @@ -610,7 +610,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, @@ -621,7 +621,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index d6ba94677ac2..58b75cf399fc 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -111,7 +111,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-play", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_init, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware1, codec, platform), @@ -120,7 +120,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-cap", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware2, codec, platform), }, diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e561464f7d60..dad70436d063 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -51,7 +51,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) /* set rt5682 dai fmt */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); + | SND_SOC_DAIFMT_CBP_CFP); if (ret < 0) { dev_err(rtd->card->dev, "Failed to set rt5682 dai fmt: %d\n", ret); @@ -302,7 +302,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = acp3x_5682_init, .dpcm_playback = 1, .dpcm_capture = 1, @@ -313,7 +313,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .dpcm_playback = 1, .ops = &acp3x_max_play_ops, .cpus = acp3x_bt, @@ -325,7 +325,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-ec-dmic0-capture", .stream_name = "Capture DMIC0", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .dpcm_capture = 1, .ops = &acp3x_ec_cap0_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), -- 2.20.1

1 1

[PATCH] ASoC: test-component: fix null pointer dereference.
by Ameer Hamza 09 Dec '21

09 Dec '21

Dereferncing of_id pointer will result in exception in current implementation since of_match_device() will assign it to NULL. Adding NULL check for protection. Signed-off-by: Ameer Hamza <amhamza.mgc(a)gmail.com> --- sound/soc/generic/test-component.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index 85385a771d80..8fc97d3ff011 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -532,13 +532,16 @@ static int test_driver_probe(struct platform_device *pdev) struct device_node *node = dev->of_node; struct device_node *ep; const struct of_device_id *of_id = of_match_device(test_of_match, &pdev->dev); - const struct test_adata *adata = of_id->data; + const struct test_adata *adata; struct snd_soc_component_driver *cdriv; struct snd_soc_dai_driver *ddriv; struct test_dai_name *dname; struct test_priv *priv; int num, ret, i; + if (!of_id) + return -EINVAL; + adata = of_id->data; num = of_graph_get_endpoint_count(node); if (!num) { dev_err(dev, "no port exits\n"); -- 2.25.1

3 14

PinePhone Pro support
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-ucm-conf issue #125 was opened from tomfitzhenry: It'd be great to have PinePhone Pro support in alsa-ucm-conf. https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/4 is another project that tracks this. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/125 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

PinePhone support
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-ucm-conf issue #124 was opened from tomfitzhenry: It'd be great to have PinePhone support in alsa-ucm-conf. https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 is the canonical source of PinePhone UCM. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/124 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-lib-1.2.6 regression, no audio
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-lib issue #199 was opened from gkubaryk: after upgrading the following packages, there is no audio: =media-plugins/alsa-plugins-1.2.6 =media-sound/alsa-utils-1.2.6 =media-libs/alsa-ucm-conf-1.2.6 =media-libs/alsa-lib-1.2.6 I checked each of the four individually and the last one (alsa-lib, 1.2.5.1) is the one that restored audio. alsa-info.sh output, as run from 1.2.5.1 (but it downloaded the newer version of the script to /tmp) is at http://alsa-project.org/db/?f=f94a60ff547f9cce9e872acae67fdf35473fa754 let me know if I can be of further assistance; I need the audio working for the rest of the day but would be able to provide output from 1.2.6 tomorrow perhaps, if desired. Issue URL : https://github.com/alsa-project/alsa-lib/issues/199 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Pacman cannot find alsa-lib=1.2.6
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-lib issue #198 was opened from TonyDaraguon: Using Arco When I try to update lib32-alsa-lib, my package manager tells me that it couldn't find the dependencies (alsa-lib=1.2.6). What should I do? Issue URL : https://github.com/alsa-project/alsa-lib/issues/198 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest
by Mark Brown 08 Dec '21

08 Dec '21

Add a basic test for the mixer control interface. For every control on every sound card in the system it checks that it can read and write the default value where the control supports that and for writeable controls attempts to write all valid values, restoring the default values after each test to minimise disruption for users. There are quite a few areas for improvement - currently no coverage of the generation of notifications, several of the control types don't have any coverage for the values and we don't have any testing of error handling when we attempt to write out of range values - but this provides some basic coverage. This is added as a kselftest since unlike other ALSA test programs it does not require either physical setup of the device or interactive monitoring by users and kselftest is one of the test suites that is frequently run by people doing general automated testing so should increase coverage. It is written in terms of alsa-lib since tinyalsa is not generally packaged for distributions which makes things harder for general users interested in kselftest as a whole but it will be a barrier to people with Android. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- v2: Use pkg-config to get CFLAGS and LDLIBS for alsa-lib. MAINTAINERS | 7 + tools/testing/selftests/Makefile | 3 +- tools/testing/selftests/alsa/.gitignore | 1 + tools/testing/selftests/alsa/Makefile | 9 + tools/testing/selftests/alsa/mixer-test.c | 616 ++++++++++++++++++++++ 5 files changed, 635 insertions(+), 1 deletion(-) create mode 100644 tools/testing/selftests/alsa/.gitignore create mode 100644 tools/testing/selftests/alsa/Makefile create mode 100644 tools/testing/selftests/alsa/mixer-test.c diff --git a/MAINTAINERS b/MAINTAINERS index 79ef55bf2ca7..ba25b33e2f96 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -17946,6 +17946,7 @@ F: Documentation/sound/ F: include/sound/ F: include/uapi/sound/ F: sound/ +F: tools/testing/selftests/alsa SOUND - COMPRESSED AUDIO M: Vinod Koul <vkoul(a)kernel.org> @@ -17965,6 +17966,12 @@ F: include/sound/dmaengine_pcm.h F: sound/core/pcm_dmaengine.c F: sound/soc/soc-generic-dmaengine-pcm.c +SOUND - ALSA SELFTESTS +M: Mark Brown <broonie(a)kernel.org> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Supported +F: tools/testing/selftests/alsa + SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> diff --git a/tools/testing/selftests/Makefile b/tools/testing/selftests/Makefile index c852eb40c4f7..d08fe4cfe811 100644 --- a/tools/testing/selftests/Makefile +++ b/tools/testing/selftests/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 -TARGETS = arm64 +TARGETS += alsa +TARGETS += arm64 TARGETS += bpf TARGETS += breakpoints TARGETS += capabilities diff --git a/tools/testing/selftests/alsa/.gitignore b/tools/testing/selftests/alsa/.gitignore new file mode 100644 index 000000000000..3bb7c41266a8 --- /dev/null +++ b/tools/testing/selftests/alsa/.gitignore @@ -0,0 +1 @@ +mixer-test diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile new file mode 100644 index 000000000000..f64d9090426d --- /dev/null +++ b/tools/testing/selftests/alsa/Makefile @@ -0,0 +1,9 @@ +# SPDX-License-Identifier: GPL-2.0 +# + +CFLAGS += $(shell pkg-config --cflags alsa) +LDLIBS += $(shell pkg-config --libs alsa) + +TEST_GEN_PROGS := mixer-test + +include ../lib.mk diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c new file mode 100644 index 000000000000..6082efa0b426 --- /dev/null +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -0,0 +1,616 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// kselftest for the ALSA mixer API +// +// Original author: Mark Brown <broonie(a)kernel.org> +// Copyright (c) 2021 Arm Limited + +// This test will iterate over all cards detected in the system, exercising +// every mixer control it can find. This may conflict with other system +// software if there is audio activity so is best run on a system with a +// minimal active userspace. + +#include <stdio.h> +#include <stdlib.h> +#include <stdbool.h> +#include <string.h> +#include <getopt.h> +#include <stdarg.h> +#include <ctype.h> +#include <math.h> +#include <errno.h> +#include <assert.h> +#include <alsa/asoundlib.h> +#include <poll.h> +#include <stdint.h> + +#include "../kselftest.h" + +#define TESTS_PER_CONTROL 3 + +struct card_data { + snd_ctl_t *handle; + int card; + int num_ctls; + snd_ctl_elem_list_t *ctls; + struct card_data *next; +}; + +struct ctl_data { + const char *name; + snd_ctl_elem_id_t *id; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *def_val; + int elem; + struct card_data *card; + struct ctl_data *next; +}; + +int num_cards = 0; +int num_controls = 0; +struct card_data *card_list = NULL; +struct ctl_data *ctl_list = NULL; + +void find_controls(void) +{ + char name[32]; + int card, ctl, err; + struct card_data *card_data; + struct ctl_data *ctl_data; + + card = -1; + if (snd_card_next(&card) < 0 || card < 0) + return; + + while (card >= 0) { + sprintf(name, "hw:%d", card); + + card_data = malloc(sizeof(*card_data)); + if (!card_data) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_open(&card_data->handle, name, 0); + if (err < 0) { + ksft_print_msg("Failed to get hctl for card %d: %s\n", + card, snd_strerror(err)); + goto next_card; + } + + /* Count controls */ + snd_ctl_elem_list_malloc(&card_data->ctls); + snd_ctl_elem_list(card_data->handle, card_data->ctls); + card_data->num_ctls = snd_ctl_elem_list_get_count(card_data->ctls); + + /* Enumerate control information */ + snd_ctl_elem_list_alloc_space(card_data->ctls, card_data->num_ctls); + snd_ctl_elem_list(card_data->handle, card_data->ctls); + + card_data->card = num_cards++; + card_data->next = card_list; + card_list = card_data; + + num_controls += card_data->num_ctls; + + for (ctl = 0; ctl < card_data->num_ctls; ctl++) { + ctl_data = malloc(sizeof(*ctl_data)); + if (!ctl_data) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + ctl_data->card = card_data; + ctl_data->elem = ctl; + ctl_data->name = snd_ctl_elem_list_get_name(card_data->ctls, + ctl); + + err = snd_ctl_elem_id_malloc(&ctl_data->id); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_elem_info_malloc(&ctl_data->info); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_elem_value_malloc(&ctl_data->def_val); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + snd_ctl_elem_list_get_id(card_data->ctls, ctl, + ctl_data->id); + snd_ctl_elem_info_set_id(ctl_data->info, ctl_data->id); + err = snd_ctl_elem_info(card_data->handle, + ctl_data->info); + if (err < 0) { + ksft_print_msg("%s getting info for %d\n", + snd_strerror(err), + ctl_data->name); + } + + snd_ctl_elem_value_set_id(ctl_data->def_val, + ctl_data->id); + + ctl_data->next = ctl_list; + ctl_list = ctl_data; + } + + next_card: + if (snd_card_next(&card) < 0) { + ksft_print_msg("snd_card_next"); + break; + } + } +} + +/* + * Check that we can read the default value and it is valid. Write + * tests use the read value to restore the default. + */ +void test_ctl_get_value(struct ctl_data *ctl) +{ + int err; + long int_val; + long long int64_val; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* Can't test reading on an unreadable control */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) { + ksft_print_msg("%s is not readable\n", ctl->name); + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + err = snd_ctl_elem_read(ctl->card->handle, ctl->def_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + goto out; + } + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_NONE: + ksft_print_msg("%s Invalid control type NONE\n", ctl->name); + err = -1; + break; + + case SND_CTL_ELEM_TYPE_BOOLEAN: + int_val = snd_ctl_elem_value_get_boolean(ctl->def_val, 0); + switch (int_val) { + case 0: + case 1: + break; + default: + ksft_print_msg("%s Invalid boolean value %ld\n", + ctl->name, int_val); + err = -1; + break; + } + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + int_val = snd_ctl_elem_value_get_integer(ctl->def_val, 0); + + if (int_val < snd_ctl_elem_info_get_min(ctl->info)) { + ksft_print_msg("%s value %ld less than minimum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_min(ctl->info)); + err = -1; + } + + if (int_val > snd_ctl_elem_info_get_max(ctl->info)) { + ksft_print_msg("%s value %ld more than maximum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_max(ctl->info)); + err = -1; + } + + /* Only check step size if there is one and we're in bounds */ + if (err >= 0 && snd_ctl_elem_info_get_step(ctl->info) && + (int_val - snd_ctl_elem_info_get_min(ctl->info) % + snd_ctl_elem_info_get_step(ctl->info))) { + ksft_print_msg("%s value %ld invalid for step %ld minimum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_step(ctl->info), + snd_ctl_elem_info_get_min(ctl->info)); + err = -1; + } + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + int64_val = snd_ctl_elem_value_get_integer64(ctl->def_val, 0); + + if (int64_val < snd_ctl_elem_info_get_min64(ctl->info)) { + ksft_print_msg("%s value %lld less than minimum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_min64(ctl->info)); + err = -1; + } + + if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) { + ksft_print_msg("%s value %lld more than maximum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_max(ctl->info)); + err = -1; + } + + /* Only check step size if there is one and we're in bounds */ + if (err >= 0 && snd_ctl_elem_info_get_step64(ctl->info) && + (int64_val - snd_ctl_elem_info_get_min64(ctl->info)) % + snd_ctl_elem_info_get_step64(ctl->info)) { + ksft_print_msg("%s value %lld invalid for step %lld minimum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_step64(ctl->info), + snd_ctl_elem_info_get_min64(ctl->info)); + err = -1; + } + break; + + default: + /* No tests for other types */ + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + +out: + ksft_test_result(err >= 0, "get_value.%d.%d\n", + ctl->card->card, ctl->elem); +} + +bool show_mismatch(struct ctl_data *ctl, int index, + snd_ctl_elem_value_t *read_val, + snd_ctl_elem_value_t *expected_val) +{ + long long expected_int, read_int; + + /* + * We factor out the code to compare values representable as + * integers, ensure that check doesn't log otherwise. + */ + expected_int = 0; + read_int = 0; + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + expected_int = snd_ctl_elem_value_get_boolean(expected_val, + index); + read_int = snd_ctl_elem_value_get_boolean(read_val, index); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + expected_int = snd_ctl_elem_value_get_integer(expected_val, + index); + read_int = snd_ctl_elem_value_get_integer(read_val, index); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + expected_int = snd_ctl_elem_value_get_integer64(expected_val, + index); + read_int = snd_ctl_elem_value_get_integer64(read_val, + index); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + expected_int = snd_ctl_elem_value_get_enumerated(expected_val, + index); + read_int = snd_ctl_elem_value_get_enumerated(read_val, + index); + break; + + default: + break; + } + + if (expected_int != read_int) { + ksft_print_msg("%s.%d expected %lld but read %lld\n", + ctl->name, index, expected_int, read_int); + return true; + } else { + return false; + } +} + +/* + * Write a value then if possible verify that we get the expected + * result. An optional expected value can be provided if we expect + * the write to fail, for verifying that invalid writes don't corrupt + * anything. + */ +int write_and_verify(struct ctl_data *ctl, + snd_ctl_elem_value_t *write_val, + snd_ctl_elem_value_t *expected_val) +{ + int err, i; + bool error_expected, mismatch_shown; + snd_ctl_elem_value_t *read_val, *w_val; + snd_ctl_elem_value_alloca(&read_val); + snd_ctl_elem_value_alloca(&w_val); + + /* + * We need to copy the write value since writing can modify + * the value which causes surprises, and allocate an expected + * value if we expect to read back what we wrote. + */ + snd_ctl_elem_value_copy(w_val, write_val); + if (expected_val) { + error_expected = true; + } else { + error_expected = false; + snd_ctl_elem_value_alloca(&expected_val); + snd_ctl_elem_value_copy(expected_val, write_val); + } + + /* + * Do the write, if we have an expected value ignore the error + * and carry on to validate the expected value. + */ + err = snd_ctl_elem_write(ctl->card->handle, w_val); + if (err < 0 && !error_expected) { + ksft_print_msg("snd_ctl_elem_write() failed: %s\n", + snd_strerror(err)); + return err; + } + + /* Can we do the verification part? */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) + return err; + + snd_ctl_elem_value_set_id(read_val, ctl->id); + + err = snd_ctl_elem_read(ctl->card->handle, read_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + return err; + } + + /* + * Use the libray to compare values, if there's a mismatch + * carry on and try to provide a more useful diagnostic than + * just "mismatch". + */ + if (!snd_ctl_elem_value_compare(expected_val, read_val)) + return 0; + + mismatch_shown = false; + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) + if (show_mismatch(ctl, i, read_val, expected_val)) + mismatch_shown = true; + + if (!mismatch_shown) + ksft_print_msg("%s read and written values differ\n", + ctl->name); + + return -1; +} + +/* + * Make sure we can write the default value back to the control, this + * should validate that at least some write works. + */ +void test_ctl_write_default(struct ctl_data *ctl) +{ + int err; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + if (!snd_ctl_elem_info_is_writable(ctl->info)) { + ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* No idea what the default was for unreadable controls */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) { + ksft_print_msg("%s couldn't read default\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + err = write_and_verify(ctl, ctl->def_val, NULL); + + ksft_test_result(err >= 0, "write_default.%d.%d\n", + ctl->card->card, ctl->elem); +} + +bool test_ctl_write_valid_boolean(struct ctl_data *ctl) +{ + int err, i, j; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = 0; j < 2; j++) { + snd_ctl_elem_value_set_boolean(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + return !fail; +} + +bool test_ctl_write_valid_integer(struct ctl_data *ctl) +{ + int err; + int i; + long j, step; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + step = snd_ctl_elem_info_get_step(ctl->info); + if (!step) + step = 1; + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = snd_ctl_elem_info_get_min(ctl->info); + j <= snd_ctl_elem_info_get_max(ctl->info); j += step) { + + snd_ctl_elem_value_set_integer(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + + return !fail; +} + +bool test_ctl_write_valid_integer64(struct ctl_data *ctl) +{ + int err, i; + long long j, step; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + step = snd_ctl_elem_info_get_step64(ctl->info); + if (!step) + step = 1; + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = snd_ctl_elem_info_get_min64(ctl->info); + j <= snd_ctl_elem_info_get_max64(ctl->info); j += step) { + + snd_ctl_elem_value_set_integer64(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + + return !fail; +} + +bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) +{ + int err, i, j; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = 0; j < snd_ctl_elem_info_get_items(ctl->info); j++) { + snd_ctl_elem_value_set_enumerated(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + return !fail; +} + +void test_ctl_write_valid(struct ctl_data *ctl) +{ + bool pass; + int err; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + if (!snd_ctl_elem_info_is_writable(ctl->info)) { + ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + pass = test_ctl_write_valid_boolean(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + pass = test_ctl_write_valid_integer(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + pass = test_ctl_write_valid_integer64(ctl); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + pass = test_ctl_write_valid_enumerated(ctl); + break; + + default: + /* No tests for this yet */ + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* Restore the default value to minimise disruption */ + err = write_and_verify(ctl, ctl->def_val, NULL); + if (err < 0) + pass = false; + + ksft_test_result(pass, "write_valid.%d.%d\n", + ctl->card->card, ctl->elem); +} + +int main(void) +{ + struct ctl_data *ctl; + + ksft_print_header(); + + find_controls(); + + ksft_set_plan(num_controls * TESTS_PER_CONTROL); + + for (ctl = ctl_list; ctl != NULL; ctl = ctl->next) { + /* + * Must test get_value() before we write anything, the + * test stores the default value for later cleanup. + */ + test_ctl_get_value(ctl); + test_ctl_write_default(ctl); + test_ctl_write_valid(ctl); + } + + ksft_exit_pass(); + + return 0; +} -- 2.30.2

6 18

[PATCH] kselftest: alsa: Use private alsa-lib configuration in mixer test
by Jaroslav Kysela 08 Dec '21

08 Dec '21

As mentined by Takashi Sakamoto, the system-wide alsa-lib configuration may override the standard device declarations. This patch use the private alsa-lib configuration to set the predictable environment. Cc: Mark Brown <broonie(a)kernel.org> Cc: Shuah Khan <shuah(a)kernel.org> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: linux-kselftest(a)vger.kernel.org Link: https://lore.kernel.org/alsa-devel/Ya7TAHdMe9i41bsC@workstation/ Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- tools/testing/selftests/alsa/mixer-test.c | 50 ++++++++++++++++++++++- 1 file changed, 49 insertions(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index b87475fb7372..0f533707484c 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -46,22 +46,68 @@ struct ctl_data { struct ctl_data *next; }; +static const char *alsa_config = +"ctl.hw {\n" +" @args [ CARD ]\n" +" @args.CARD.type string\n" +" type hw\n" +" card $CARD\n" +"}\n" +; + int num_cards = 0; int num_controls = 0; struct card_data *card_list = NULL; struct ctl_data *ctl_list = NULL; +#if !defined(SND_LIB_VER) || SND_LIB_VERSION < SND_LIB_VER(1, 2, 6) +int snd_config_load_string(snd_config_t **config, const char *s, size_t size) +{ + snd_input_t *input; + snd_config_t *dst; + int err; + + assert(config && s); + if (size == 0) + size = strlen(s); + err = snd_input_buffer_open(&input, s, size); + if (err < 0) + return err; + err = snd_config_top(&dst); + if (err < 0) { + snd_input_close(input); + return err; + } + err = snd_config_load(dst, input); + snd_input_close(input); + if (err < 0) { + snd_config_delete(dst); + return err; + } + *config = dst; + return 0; +} +#endif + void find_controls(void) { char name[32]; int card, ctl, err; struct card_data *card_data; struct ctl_data *ctl_data; + snd_config_t *config; card = -1; if (snd_card_next(&card) < 0 || card < 0) return; + err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); + if (err < 0) { + ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", + snd_strerror(err)); + ksft_exit_fail(); + } + while (card >= 0) { sprintf(name, "hw:%d", card); @@ -71,7 +117,7 @@ void find_controls(void) ksft_exit_fail(); } - err = snd_ctl_open(&card_data->handle, name, 0); + err = snd_ctl_open_lconf(&card_data->handle, name, 0, config); if (err < 0) { ksft_print_msg("Failed to get hctl for card %d: %s\n", card, snd_strerror(err)); @@ -147,6 +193,8 @@ void find_controls(void) break; } } + + snd_config_delete(config); } /* -- 2.31.1

3 5

vxloader: no VX-compatible cards found VX222v2
by GitHub issues - opened 08 Dec '21

08 Dec '21

alsa-project/alsa-tools issue #9 was opened from LGCW: I have an AMD AM4 motherboard with two legacy PCI slots. The bridge uses the ASMedia ASM1083. Running vxloader produces the following error. ``` vxloader vxloader: no VX-compatible cards found vxloader -D hw:0 vxloader: cannot open hwdep hw:0 ``` If it's anything like the 9632 the firmware has to be loaded in order to function? Best I can tell Debian 11 sees the card. ``` lsmod | grep vx snd_vx222 24576 0 snd_vx_lib 57344 1 snd_vx222 snd_pcm 147456 2 snd_hdsp,snd_vx_lib snd 110592 8 snd_seq_device,snd_hwdep,snd_hdsp,snd_timer,snd_vx_lib,snd_vx222,snd_pcm,snd_rawmidi ``` ``` cat /proc/asound/card0/vx-status Digigram VX222/v2 at 0xf100 & 0xf000, irq 25 Xilinx Firmware: Loaded Device Initialized: Yes DSP audio info: realtime linear8 linear16 linear24 Input Source: Analog Clock Mode: Internal Clock Source: Internal Frequency: 48000 Detected Frequency: 0 Detected UER type: Not Present Min/Max/Cur IBL: 252/2016/252 (granularity=126) ``` I noticed hdsploader picked it up as well. ``` hdsploader hdsploader - firmware loader for RME Hammerfall DSP cards Looking for HDSP + Multiface or Digiface cards : Card 0 : Digigram VX222/v2 at 0xf100 & 0xf000, irq 25 Card 1 : RME Hammerfall HDSP 9632 at 0xf7400000, irq 27 ``` Issue URL : https://github.com/alsa-project/alsa-tools/issues/9 Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH v2 0/1] ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output
by Ajit Kumar Pandey 08 Dec '21

08 Dec '21

changes since v1: - Fix typo in commit description Ajit Kumar Pandey (1): ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output sound/soc/codecs/rt5682s.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) -- 2.25.1

2 2

[PATCH] ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output
by Ajit Kumar Pandey 08 Dec '21

08 Dec '21

Add dapm switch in playback path to mute or unmute HP output data. We will set and reset MUTE_SFT bit in RT5682S_HP_CTRL_1 register based on switch value to mute or umute respective channel. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/codecs/rt5682s.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d49a4f68566d..21ab5f7df422 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1573,6 +1573,14 @@ static const char * const rt5682s_adcdat_pin_select[] = { "ADCDAT1", "ADCDAT2", }; +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, + RT5682S_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, + RT5682S_R_MUTE_SFT, 1, 1); + static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_adcdat_pin_enum, RT5682S_GPIO_CTRL_1, RT5682S_GP4_PIN_SFT, RT5682S_GP4_PIN_MASK, rt5682s_adcdat_pin_select, rt5682s_adcdat_pin_values); @@ -1746,6 +1754,11 @@ static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682s_hp_amp_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + /* CLK DET */ SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, RT5682S_SYS_CLK_DET_SFT, 0, NULL, 0), @@ -1895,8 +1908,10 @@ static const struct snd_soc_dapm_route rt5682s_dapm_routes[] = { {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "SAR"}, - {"HPOL", NULL, "HP Amp"}, - {"HPOR", NULL, "HP Amp"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, }; static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, -- 2.25.1

1 0

[PATCH] ASoC: mediatek: mt8195: silence uninitialized variable warning
by Dan Carpenter 08 Dec '21

08 Dec '21

Smatch complains that we might hit the continue path on every iteration through the loop. sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c:831 mt8195_mt6359_rt1019_rt5682_card_late_probe() error: uninitialized symbol 'sof_comp'. Initialize "sof_comp" to NULL to silence this warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index b240610dcef0..11a185da0d96 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -813,7 +813,7 @@ static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8195_mt6359_rt1019_rt5682_card_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *runtime; - struct snd_soc_component *sof_comp; + struct snd_soc_component *sof_comp = NULL; int i; /* 1. find sof component */ -- 2.20.1

2 1

[PATCH] ASoC: AMD: acp-config: fix missing dependency on SND_SOC_ACPI
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

With a custom .config, the following error is thrown: ERROR: modpost: "snd_soc_acpi_codec_list" [sound/soc/amd/snd-acp-config.ko] undefined! Fix by adding a clear dependency on SND_SOC_ACPI Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 8961b8fd23eb..bcfeb3fc2592 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_AMD_YC_MACH config SND_AMD_ACP_CONFIG tristate "AMD ACP configuration selection" + depends on SND_SOC_ACPI help This option adds an auto detection to determine which ACP driver modules to use -- 2.25.1

2 1

[PATCH 2/2] ASoC: rt5682s: add delay time to fix pop sound issue
by shumingf＠realtek.com 08 Dec '21

08 Dec '21

From: Shuming Fan <shumingf(a)realtek.com> There is a pop noise at the beginning of the capture data. This patch adds the delay time before stereo1 ADC unmute to fix the pop sound issue. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- include/sound/rt5682s.h | 1 + sound/soc/codecs/rt5682s.c | 34 +++++++++++++++++++++++++++++++--- 2 files changed, 32 insertions(+), 3 deletions(-) diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h index accfbc2dcdd2..f18d91308b9a 100644 --- a/include/sound/rt5682s.h +++ b/include/sound/rt5682s.h @@ -40,6 +40,7 @@ struct rt5682s_platform_data { enum rt5682s_jd_src jd_src; unsigned int dmic_clk_rate; unsigned int dmic_delay; + unsigned int amic_delay; bool dmic_clk_driving_high; const char *dai_clk_names[RT5682S_DAI_NUM_CLKS]; diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d49a4f68566d..efa1016831dd 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1367,6 +1367,31 @@ static int rt5682s_hp_amp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5682s_stereo1_adc_mixl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int delay = 0; + + if (rt5682s->pdata.amic_delay) + delay = rt5682s->pdata.amic_delay; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(delay); + snd_soc_component_update_bits(component, RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE, RT5682S_L_MUTE); + break; + } + + return 0; +} + static int sar_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1680,9 +1705,10 @@ static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { /* ADC Mixer */ SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682S_PWR_DIG_2, RT5682S_PWR_ADC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), - SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682S_STO1_ADC_DIG_VOL, - RT5682S_L_MUTE_SFT, 1, rt5682s_sto1_adc_l_mix, - ARRAY_SIZE(rt5682s_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER_E("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_adc_l_mix, ARRAY_SIZE(rt5682s_sto1_adc_l_mix), + rt5682s_stereo1_adc_mixl_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682S_STO1_ADC_DIG_VOL, RT5682S_R_MUTE_SFT, 1, rt5682s_sto1_adc_r_mix, ARRAY_SIZE(rt5682s_sto1_adc_r_mix)), @@ -2885,6 +2911,8 @@ static int rt5682s_parse_dt(struct rt5682s_priv *rt5682s, struct device *dev) &rt5682s->pdata.dmic_clk_rate); device_property_read_u32(dev, "realtek,dmic-delay-ms", &rt5682s->pdata.dmic_delay); + device_property_read_u32(dev, "realtek,amic-delay-ms", + &rt5682s->pdata.amic_delay); rt5682s->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); -- 2.34.1

2 1

[PATCH 1/2] ASoC: dt-bindings: rt5682s: add AMIC delay time property
by shumingf＠realtek.com 08 Dec '21

08 Dec '21

From: Shuming Fan <shumingf(a)realtek.com> Add the AMIC delay time to control how much delay time (ms) to unmute the stereo1 ADC. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml index 2b8b7b51fe55..d65c0ed5060c 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -61,6 +61,10 @@ properties: description: | Set the delay time (ms) for the requirement of the particular DMIC. + realtek,amic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular platform or AMIC. + realtek,dmic-clk-driving-high: type: boolean description: | -- 2.34.1

2 1

[PATCH] ASoC: Intel: boards: add 'static' qualifiers for max98390 routes
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

Sparse warnings: sound/soc/intel/boards/sof_maxim_common.c:140:33: error: symbol 'max_98390_dapm_routes' was not declared. Should it be static? sound/soc/intel/boards/sof_maxim_common.c:156:33: error: symbol 'max_98390_tt_dapm_routes' was not declared. Should it be static? Fixes: f316c9d9ba8ea ('ASoC: Intel: boards: add max98390 2/4 speakers support') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(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 9171d9cd179e..112e89951da0 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -137,7 +137,7 @@ EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); /* * Maxim MAX98390 */ -const struct snd_soc_dapm_route max_98390_dapm_routes[] = { +static const struct snd_soc_dapm_route max_98390_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "Left BE_OUT" }, { "Right Spk", NULL, "Right BE_OUT" }, @@ -153,7 +153,7 @@ static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { SND_SOC_DAPM_SPK("TR Spk", NULL), }; -const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { +static const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { /* Tweeter speaker */ { "TL Spk", NULL, "Tweeter Left BE_OUT" }, { "TR Spk", NULL, "Tweeter Right BE_OUT" }, -- 2.25.1

2 1

[PATCH] ASoC: Intel: sof_rt5682: Move rt1015 speaker amp to common file
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

From: Yong Zhi <yong.zhi(a)intel.com> Move rt1015 driver code to common file to be consistent with rt1011 and rt1015p. No functional change. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_realtek_common.c | 119 +++++++++++++++++++- sound/soc/intel/boards/sof_realtek_common.h | 7 ++ sound/soc/intel/boards/sof_rt5682.c | 107 +----------------- 3 files changed, 127 insertions(+), 106 deletions(-) diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 2ec34f8df9e1..4cf131310ad3 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -12,12 +12,13 @@ #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> #include "../../codecs/rt1011.h" +#include "../../codecs/rt1015.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[] = { +static const struct snd_soc_dapm_route speaker_map_lr[] = { /* speaker */ { "Left Spk", NULL, "Left SPO" }, { "Right Spk", NULL, "Right SPO" }, @@ -117,8 +118,8 @@ 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)); + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); if (ret) dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); return ret; @@ -241,3 +242,115 @@ void sof_rt1015p_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1015p_codec_confs; card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); } + +/* + * RT1015 audio amplifier + */ + +static int rt1015_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 i, fs = 64, ret; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * fs, + params_rate(params) * 256); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + + return 0; +} + +static int rt1015_hw_params_pll_and_tdm(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 i, fs = 100, ret; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * fs, + params_rate(params) * 256); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + /* rx slot 1 for RT1015_DEV0_NAME */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), + 0x0, 0x1, 4, 24); + if (ret) + return ret; + + /* rx slot 2 for RT1015_DEV1_NAME */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 1), + 0x0, 0x2, 4, 24); + if (ret) + return ret; + + return 0; +} + +static struct snd_soc_ops rt1015_ops = { + .hw_params = rt1015_hw_params, +}; + +static struct snd_soc_codec_conf rt1015_amp_conf[] = { + { + .dlc = COMP_CODEC_CONF(RT1015_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015_components[] = { + { + .name = RT1015_DEV0_NAME, + .dai_name = RT1015_CODEC_DAI, + }, + { + .name = RT1015_DEV1_NAME, + .dai_name = RT1015_CODEC_DAI, + }, +}; + +static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) +{ + return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); +} + +void sof_rt1015_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1015_amp_conf; + card->num_configs = ARRAY_SIZE(rt1015_amp_conf); +} + +void sof_rt1015_dai_link(struct snd_soc_dai_link *link, unsigned int fs) +{ + link->codecs = rt1015_components; + link->num_codecs = ARRAY_SIZE(rt1015_components); + link->init = speaker_codec_init_lr; + link->ops = &rt1015_ops; + + if (fs == 100) + rt1015_ops.hw_params = rt1015_hw_params_pll_and_tdm; +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index cb0b49b2855c..228ac9c08430 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -28,4 +28,11 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card); void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); void sof_rt1015p_codec_conf(struct snd_soc_card *card); +#define RT1015_CODEC_DAI "rt1015-aif" +#define RT1015_DEV0_NAME "i2c-10EC1015:00" +#define RT1015_DEV1_NAME "i2c-10EC1015:01" + +void sof_rt1015_dai_link(struct snd_soc_dai_link *link, unsigned int fs); +void sof_rt1015_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 6cadb5fb72e0..bd6d2e7dea53 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -20,7 +20,6 @@ #include <sound/rt5682.h> #include <sound/rt5682s.h> #include <sound/soc-acpi.h> -#include "../../codecs/rt1015.h" #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" #include "../../codecs/hdac_hdmi.h" @@ -400,67 +399,6 @@ static struct snd_soc_ops sof_rt5682_ops = { .hw_params = sof_rt5682_hw_params, }; -static int sof_rt1015_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_card *card = rtd->card; - struct snd_soc_dai *codec_dai; - int i, fs, ret; - - if (!snd_soc_card_get_codec_dai(card, "rt1015-aif")) - return 0; - - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_100FS) - fs = 100; - else - fs = 64; - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, - params_rate(params) * fs, - params_rate(params) * 256); - if (ret < 0) { - dev_err(card->dev, "failed to set pll\n"); - return ret; - } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, - params_rate(params) * 256, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "failed to set sysclk\n"); - return ret; - } - - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_100FS) { - if (!strcmp(codec_dai->component->name, "i2c-10EC1015:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x0, 0x1, 4, 24); - if (ret < 0) { - dev_err(card->dev, "failed to set tdm slot\n"); - return ret; - } - } - - if (!strcmp(codec_dai->component->name, "i2c-10EC1015:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x0, 0x2, 4, 24); - if (ret < 0) { - dev_err(card->dev, "failed to set tdm slot\n"); - return ret; - } - } - } - } - - return 0; -} - -static struct snd_soc_ops sof_rt1015_ops = { - .hw_params = sof_rt1015_hw_params, -}; - static struct snd_soc_dai_link_component platform_component[] = { { /* name might be overridden during probe */ @@ -551,22 +489,11 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; -static const struct snd_soc_dapm_route speaker_map_lr[] = { - { "Left Spk", NULL, "Left SPO" }, - { "Right Spk", NULL, "Right SPO" }, -}; - static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; -static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) -{ - return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, - ARRAY_SIZE(speaker_map_lr)); -} - static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -589,17 +516,6 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static struct snd_soc_codec_conf rt1015_amp_conf[] = { - { - .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"), - .name_prefix = "Left", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"), - .name_prefix = "Right", - }, -}; - /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { .name = "rt5682", /* the sof- prefix is added by the core */ @@ -635,17 +551,6 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; -static struct snd_soc_dai_link_component rt1015_components[] = { - { - .name = "i2c-10EC1015:00", - .dai_name = "rt1015-aif", - }, - { - .name = "i2c-10EC1015:01", - .dai_name = "rt1015-aif", - }, -}; - static struct snd_soc_dai_link_component dummy_component[] = { { .name = "snd-soc-dummy", @@ -798,10 +703,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].id = id; if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { - links[id].codecs = rt1015_components; - links[id].num_codecs = ARRAY_SIZE(rt1015_components); - links[id].init = speaker_codec_init_lr; - links[id].ops = &sof_rt1015_ops; + sof_rt1015_dai_link(&links[id], (sof_rt5682_quirk & + SOF_RT1015_SPEAKER_AMP_100FS) ? 100 : 64); } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & @@ -995,10 +898,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_rt5682.dai_link = dai_links; - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { - sof_audio_card_rt5682.codec_conf = rt1015_amp_conf; - sof_audio_card_rt5682.num_configs = ARRAY_SIZE(rt1015_amp_conf); - } + if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) + sof_rt1015_codec_conf(&sof_audio_card_rt5682); INIT_LIST_HEAD(&ctx->hdmi_pcm_list); -- 2.25.1

2 1

[PATCH] sound: fsl: add missing put_device() call in imx_hdmi_probe()
by Qing Wang 08 Dec '21

08 Dec '21

From: Wang Qing <wangqing(a)vivo.com> of_find_device_by_node() takes a reference to the embedded struct device which needs to be dropped when error return. Signed-off-by: Wang Qing <wangqing(a)vivo.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 929f69b..89fd8ad --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -125,6 +125,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { + put_device(&cpu_pdev->dev); ret = -ENOMEM; goto fail; } -- 2.7.4

2 1

Re: [Linux-stm32] [PATCH v2 1/4] ASoC: dt-bindings: stm32: i2s: add audio-graph-card port
by Alexandre TORGUE 08 Dec '21

08 Dec '21

Hi Ahmad On 12/7/21 2:59 PM, Ahmad Fatoum wrote: > Hello Alex, > > On 07.12.21 14:52, Alexandre TORGUE wrote: >> Hi Rob >> >> On 12/1/21 11:34 PM, Rob Herring wrote: >>> On Fri, Nov 26, 2021 at 11:25:27AM +0100, Olivier MOYSAN wrote: >>>> Hi Rob, >>>> >>>> On 11/25/21 10:26 PM, Rob Herring wrote: >>>>> On Thu, 25 Nov 2021 15:40:50 +0100, Olivier Moysan wrote: >>>>>> The STM2 I2S DAI can be connected via the audio-graph-card. >>>>>> Add port entry into the bindings. >>>>>> >>>>>> Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> >>>>>> --- >>>>>> Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 5 +++++ >>>>>> 1 file changed, 5 insertions(+) >>>>>> >>>>> >>>>> Running 'make dtbs_check' with the schema in this patch gives the >>>>> following warnings. Consider if they are expected or the schema is >>>>> incorrect. These may not be new warnings. >>>>> >>>>> Note that it is not yet a requirement to have 0 warnings for dtbs_check. >>>>> This will change in the future. >>>>> >>>>> Full log is available here: https://patchwork.ozlabs.org/patch/1559750 >>>>> >>>>> >>>>> audio-controller@4000b000: 'port' does not match any of the regexes: '^port@[0-9]', 'pinctrl-[0-9]+' >>>>> arch/arm/boot/dts/stm32mp157a-dk1.dt.yaml >>>>> arch/arm/boot/dts/stm32mp157c-dk2.dt.yaml >>>>> >>>> >>>> This warning is not a new one. >>>> >>>> The i2s2 node in stm32mp15xx-dkx.dtsi would require the following binding: >>>> port: >>>> $ref: audio-graph-port.yaml# >>>> unevaluatedProperties: false >>>> >>>> However the spi binding requires to introduce a unit address: >>>> patternProperties: >>>> '^port@[0-9]': >>>> $ref: audio-graph-port.yaml# >>>> unevaluatedProperties: false >>>> >>>> The warning can be removed by re-ordering the bindings patches in the serie, >>>> as "additionalProperties: true" makes the check more tolerant on extra >>>> properties. >>> >>> That's never right. >>> >>>> The patch "ASoC: dt-bindings: stm32: i2s: add audio-graph-card port" can >>>> even be merely dropped. >>>> So, I suggest to resend the serie without audio-graph-card patch. >>> >>> Only if you aren't using audio-graph-card. >>> >>>> >>>> Does it sound too permissive to you ? >>> >>> I think perhaps you need to combine the schemas into 1. Or you need to >>> restructure your dtsi files such that you only add spi specific >>> properties when spi mode is enabled and only add i2s specific properties >>> when i2s mode is enabled. Or use the /delete-property/ directive. >> >> Initially the aim of this series was to fix a "make W=1" warnings seen on spi and i2s nodes (duplicate unit-address). Moving both nodes in a common node + using a different compatible depending on SPI or I2S usage sounded good) but it is not enough. In this series the common node is named as following: "spi2s2: spi@4000b000". It is fine for a spi usage but if we want to use this "common node" with I2S compatible and specific bindings, the node name remains spi@... and then specific spi checks are done. For this with this series applied we got this issue reported by spi-controller.yaml: >> >> spi@4000b000: port@0: 'compatible' is a required property >> >> So, if we use two separates nodes we got W=1 warning and if we use a common node we got yaml check issue. One possibility would be to use a common node with a new node name (for example i2spi@...) but I think it is not acceptable. >> >> How to progress ? > > Atmel Flexcom can be configured to be either UART, SPI or i2c. Functions > are child nodes of the flexcom node and the MFD driver matching against it, > just configure the operating mode and then calls of_platform_populate. > > Would something along these lines fit here as well? Yes it could but in my mind it was not a MFD as both feature cannot be used at the same time: it is either SPI or I2S and choice is done "statically" in device tree depending board usage. Lee, what it is your feeling about that ? Will you accept to add a MFD driver for this SPI/I2S peripheral whose prurpose is only to populate child node (either SPI or I2S) ? Cheers Alex > > Cheers, > Ahmad > >> >> Thanks >> Alex >> >> >>> Rob >>> >> >> _______________________________________________ >> Linux-stm32 mailing list >> Linux-stm32(a)st-md-mailman.stormreply.com >> https://st-md-mailman.stormreply.com/mailman/listinfo/linux-stm32 >> > >

2 2

[PATCH 1/1] ASoC: dt-bindings: audio-graph-port: enable both flag/phandle for bitclock/frame-master
by Kuninori Morimoto 08 Dec '21

08 Dec '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_daifmt_parse_clock_provider_raw() is handling both bitclock/frame-master, and is supporting both flag/phandle. Current DT is assuming it is flag style. This patch allows both case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/audio-graph-port.yaml | 23 +++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 43e7f86e3b23..6c3645a0d36b 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -14,6 +14,24 @@ select: false allOf: - $ref: /schemas/graph.yaml#/$defs/port-base +definitions: + flag_or_phandle: + oneOf: + - type: boolean + const: true + - type: 'null' + - type: array + minItems: 1 + maxItems: 1 + items: + type: array + items: + phandle: true + type: integer + maximum: 0xffffffff + minItems: 1 + maxItems: 1 + properties: prefix: description: "device name prefix" @@ -42,10 +60,11 @@ patternProperties: $ref: /schemas/types.yaml#/definitions/flag frame-master: description: Indicates dai-link frame master. - $ref: /schemas/types.yaml#/definitions/phandle + $ref: "#/definitions/flag_or_phandle" bitclock-master: description: Indicates dai-link bit clock master - $ref: /schemas/types.yaml#/definitions/phandle + $ref: "#/definitions/flag_or_phandle" + dai-format: description: audio format. items: -- 2.25.1

1 1

topology: Add option to pass include path for conditional includes
by GitHub pull_request - opened 08 Dec '21

08 Dec '21

alsa-project/alsa-utils pull request #125 was opened from ranj063: The include path passed with -I option will override the relative include path based on the source file. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/125 Patch URL : https://github.com/alsa-project/alsa-utils/pull/125.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Feature request: ACPI NHLT generation for topology2
by GitHub issues - opened 07 Dec '21

07 Dec '21

alsa-project/alsa-utils issue #124 was opened from juimonen: I'd like to add ACPI NHLT generation into topology2 parsing. Why: In certain future sof firmware versions this is needed for DAI configuration Why to alsa-utils: adding it to kernel driver was internally rejected, also I can't link to topology2 parsing outside alsa-utils Good thing: if you ever wondered how acpi nhlt blobs are created and what they contain in intel platforms -> this code is doing it. Bad things: I can think many, but I'm trying to do this with least harm to every party (kernel driver, firmware) Example of the code can be found here: https://github.com/thesofproject/alsa-utils/pull/4 So mainly asking for comments/suggestions from @perexg @lrgirdwo @ranj063 FYI. Issue URL : https://github.com/alsa-project/alsa-utils/issues/124 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RESEND, PATCH v6 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 07 Dec '21

07 Dec '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..3b3fa43fa961 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 2

[PATCH] ALSA: seq: Set upper limit of processed events
by Takashi Iwai 07 Dec '21

07 Dec '21

Currently ALSA sequencer core tries to process the queued events as much as possible when they become dispatchable. If applications try to queue too massive events to be processed at the very same timing, the sequencer core would still try to process such all events, either in the interrupt context or via some notifier; in either away, it might be a cause of RCU stall or such problems. As a potential workaround for those problems, this patch adds the upper limit of the amount of events to be processed. The remaining events are processed in the next batch, so they won't be lost. For the time being, it's limited up to 1000 events per queue, which should be high enough for any normal usages. Reported-by: Zqiang <qiang.zhang1211(a)gmail.com> Reported-by: syzbot+bb950e68b400ab4f65f8(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20211102033222.3849-1-qiang.zhang1211@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index d6c02dea976c..bc933104c3ee 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -235,12 +235,15 @@ struct snd_seq_queue *snd_seq_queue_find_name(char *name) /* -------------------------------------------------------- */ +#define MAX_CELL_PROCESSES_IN_QUEUE 1000 + void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) { unsigned long flags; struct snd_seq_event_cell *cell; snd_seq_tick_time_t cur_tick; snd_seq_real_time_t cur_time; + int processed = 0; if (q == NULL) return; @@ -263,6 +266,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } /* Process time queue... */ @@ -272,14 +277,19 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } + out: /* free lock */ spin_lock_irqsave(&q->check_lock, flags); if (q->check_again) { q->check_again = 0; - spin_unlock_irqrestore(&q->check_lock, flags); - goto __again; + if (processed < MAX_CELL_PROCESSES_IN_QUEUE) { + spin_unlock_irqrestore(&q->check_lock, flags); + goto __again; + } } q->check_blocked = 0; spin_unlock_irqrestore(&q->check_lock, flags); -- 2.31.1

1 0

[PATCH] ALSA: mixart: Add sanity check for timer notify streams
by Takashi Iwai 07 Dec '21

07 Dec '21

The miXart timer notification is a variable length, and if a hardware is screwed up, we may access over the actual data size. Let's add a sanity check and bail out if an invalid value is received. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/mixart/mixart_core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 853083dd4bad..a047ed0f84e9 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -443,6 +443,8 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) notify = (struct mixart_timer_notify *)mixart_msg_data; BUILD_BUG_ON(sizeof(notify) > sizeof(mixart_msg_data)); + if (snd_BUG_ON(notify->stream_count > ARRAY_SIZE(notify->streams))) + break; for(i=0; i<notify->stream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; -- 2.31.1

1 0

[PATCH v4 0/5] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 07 Dec '21

07 Dec '21

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (5): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Move chip specific functions to right files pinctrl: qcom: Update clock voting as optional pinctrl: qcom: Add SC7280 lpass pin configuration .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 130 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 130 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 252 +-------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 98 ++++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 169 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 166 ++++++++++++++ 9 files changed, 704 insertions(+), 374 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 13

[PATCH] ASoC: soc-core: add the driver component name to the component struc
by Kory Maincent 07 Dec '21

07 Dec '21

The function fmt_single_name is using to sanitize namings, and it return the name set to component->name. This function is returning either the "device_name" or the "device_name.driver_name" for i2c devices. There is no use of the component driver name. If a non i2c driver register two components the function will return the same "device_name" for both components. This could cause unexpected issue, in my case it is a debugfs error which tries to create two directory with the same component name. I have fixed it by prefixing the component name with the driver component name. Signed-off-by: Kory Maincent <kory.maincent(a)bootlin.com> --- sound/soc/soc-core.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcf6be4c4aaa..21ff77b231b8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2342,10 +2342,10 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); * Simplify DAI link configuration by removing ".-1" from device names * and sanitizing names. */ -static char *fmt_single_name(struct device *dev, int *id) +static char *fmt_single_name(struct device *dev, const char *snd_drv_name, int *id) { const char *devname = dev_name(dev); - char *found, *name; + char *found, *name, *tmp; unsigned int id1, id2; if (devname == NULL) @@ -2380,6 +2380,14 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } + if (snd_drv_name != NULL) { + /* Add driver component name if present */ + tmp = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", snd_drv_name, name); + devm_kfree(dev, name); + name = devm_kstrdup(dev, tmp, GFP_KERNEL); + devm_kfree(dev, tmp); + } + return name; } @@ -2444,7 +2452,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, */ if (legacy_dai_naming && (dai_drv->id == 0 || dai_drv->name == NULL)) { - dai->name = fmt_single_name(dev, &dai->id); + dai->name = fmt_single_name(dev, dai_drv->name, &dai->id); } else { dai->name = fmt_multiple_name(dev, dai_drv); if (dai_drv->id) @@ -2578,7 +2586,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, driver->name, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.25.1

2 4

[PATCH] sound: ppc: beep: fix clang -Wimplicit-fallthrough
by Anders Roxell 07 Dec '21

07 Dec '21

Clang warns: sound/ppc/beep.c:103:2: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough] case SND_TONE: break; ^ sound/ppc/beep.c:103:2: note: insert 'break;' to avoid fall-through case SND_TONE: break; ^ break; 1 warning generated. Clang is more pedantic than GCC, which does not warn when failing through to a case that is just break or return. Clang's version is more in line with the kernel's own stance in deprecated.rst. Add athe missing break to silence the warning. Reported-by: Naresh Kamboju <naresh.kamboju(a)linaro.org> Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/ppc/beep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index 0f4bce1c0d4f..bf289783eafd 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -99,7 +99,7 @@ static int snd_pmac_beep_event(struct input_dev *dev, unsigned int type, return -1; switch (code) { - case SND_BELL: if (hz) hz = 1000; + case SND_BELL: if (hz) hz = 1000; break; case SND_TONE: break; default: return -1; } -- 2.33.0

2 1

Logitech Z10 USB speakers need a volume change before audio works
by Hans de Goede 07 Dec '21

07 Dec '21

Hi, I've a set of Logitech Z10 USB speakers, which act as a USB soundcard. They have this weird glitch where after turning off my PC (and their power-supply as well) and then turning things back on, they are silent until I change the PCM volume control for the speakers inside alsa-mixer. It seems like they need some "set-volume" command to be send over the USB bus to unmute them when initially powered-up / turned on. Is their some existing usb-audio quirk which I can try to work around this? Regards, Hans

2 8

Add headset Mic supported for Lenovo platform
by Kailang 07 Dec '21

07 Dec '21

Hi Takashi, Attach patch will enable headset Mic support on Lenovo platform. Many Thanks. BR, Kailang

2 1

[PATCH] ALSA: mixart: Reduce size of mixart_timer_notify
by Kees Cook 07 Dec '21

07 Dec '21

The mixart_timer_notify structure was larger than could be represented by the mixart_msg_data array storage. Adjust the size to as large as possible to fix the warning seen with -Warray-bounds builds: sound/pci/mixart/mixart_core.c: In function 'snd_mixart_threaded_irq': sound/pci/mixart/mixart_core.c:447:50: error: array subscript 'struct mixart_timer_notify[0]' is partly outside array bounds of 'u32[128]' {aka 'unsigned int[128]'} [-Werror=array-bounds] 447 | for(i=0; i<notify->stream_count; i++) { | ^~ sound/pci/mixart/mixart_core.c:328:12: note: while referencing 'mixart_msg_data' 328 | static u32 mixart_msg_data[MSG_DEFAULT_SIZE / 4]; | ^~~~~~~~~~~~~~~ Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/pci/mixart/mixart_core.c | 3 +-- sound/pci/mixart/mixart_core.h | 10 +++++++++- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index fb8895af0363..853083dd4bad 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -23,8 +23,6 @@ #define MSG_DESCRIPTOR_SIZE 0x24 #define MSG_HEADER_SIZE (MSG_DESCRIPTOR_SIZE + 4) -#define MSG_DEFAULT_SIZE 512 - #define MSG_TYPE_MASK 0x00000003 /* mask for following types */ #define MSG_TYPE_NOTIFY 0 /* embedded -> driver (only notification, do not get_msg() !) */ #define MSG_TYPE_COMMAND 1 /* driver <-> embedded (a command has no answer) */ @@ -444,6 +442,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) struct mixart_timer_notify *notify; notify = (struct mixart_timer_notify *)mixart_msg_data; + BUILD_BUG_ON(sizeof(notify) > sizeof(mixart_msg_data)); for(i=0; i<notify->stream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; diff --git a/sound/pci/mixart/mixart_core.h b/sound/pci/mixart/mixart_core.h index fbf4731a276d..2f0e29ed5d63 100644 --- a/sound/pci/mixart/mixart_core.h +++ b/sound/pci/mixart/mixart_core.h @@ -49,6 +49,7 @@ enum mixart_message_id { MSG_CLOCK_SET_PROPERTIES = 0x200002, }; +#define MSG_DEFAULT_SIZE 512 struct mixart_msg { @@ -251,10 +252,17 @@ struct mixart_sample_pos u32 sample_pos_low_part; } __attribute__((packed)); +/* + * This structure is limited by the size of MSG_DEFAULT_SIZE. Instead of + * having MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS many streams, + * this is capped to have a total size below MSG_DEFAULT_SIZE. + */ +#define MIXART_MAX_TIMER_NOTIFY_STREAMS \ + ((MSG_DEFAULT_SIZE - sizeof(u32)) / sizeof(struct mixart_sample_pos)) struct mixart_timer_notify { u32 stream_count; - struct mixart_sample_pos streams[MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS]; + struct mixart_sample_pos streams[MIXART_MAX_TIMER_NOTIFY_STREAMS]; } __attribute__((packed)); -- 2.30.2

2 1

[GIT PULL] ASoC fixes for v5.16-rc4
by Mark Brown 07 Dec '21

07 Dec '21

The following changes since commit 86f74ba3fef56dd1cee19b7a15ae27fc0da5bb61: ASoC: SOF: hda: reset DAI widget before reconfiguring it (2021-11-24 12:57:11 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.16-rc4 for you to fetch changes up to 3fc27e9a1f619b50700f020e6cd270c1b74755f0: ASoC: codecs: wsa881x: fix return values from kcontrol put (2021-12-01 14:13:53 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.16 A relatively large collection of updates, the size is increased quite a bit by there being some repetitive changes for similar issues that occur multiple times with both notifying control value changes and runtime PM. The Rockchip update looks at first glance like a cleanup but fixes instantiation of the hardware on some systems. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: amd: fix uninitialized variable in snd_acp6x_probe() David Heidelberg (1): ASoC: dt-bindings: wlf,wm8962: add missing interrupt property Hui Wang (1): ASoC: SOF: Intel: Retry codec probing if it fails Mark Brown (1): Suspend related fixes on Tegra Nicolas Frattaroli (1): ASoC: rockchip: i2s_tdm: Dup static DAI template Rob Clark (2): ASoC: rt5682: Fix crash due to out of scope stack vars ASoC: rt5682s: Fix crash due to out of scope stack vars Sameer Pujar (6): ASoC: tegra: Balance runtime PM count ASoC: tegra: Use normal system sleep for SFC ASoC: tegra: Use normal system sleep for MVC ASoC: tegra: Use normal system sleep for Mixer ASoC: tegra: Use normal system sleep for AMX ASoC: tegra: Use normal system sleep for ADX Srinivas Kandagatla (4): ASoC: qdsp6: q6routing: Fix return value from msm_routing_put_audio_mixer ASoC: codecs: wcd934x: handle channel mappping list correctly ASoC: codecs: wcd934x: return correct value from mixer put ASoC: codecs: wsa881x: fix return values from kcontrol put .../devicetree/bindings/sound/wlf,wm8962.yaml | 3 + sound/soc/amd/yc/pci-acp6x.c | 3 +- sound/soc/codecs/rt5682.c | 10 +- sound/soc/codecs/rt5682s.c | 10 +- sound/soc/codecs/wcd934x.c | 126 +++++++++++++++------ sound/soc/codecs/wsa881x.c | 16 ++- sound/soc/qcom/qdsp6/q6routing.c | 8 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 52 +++++---- sound/soc/sof/intel/hda-codec.c | 14 ++- sound/soc/tegra/tegra210_adx.c | 4 +- sound/soc/tegra/tegra210_amx.c | 4 +- sound/soc/tegra/tegra210_mixer.c | 4 +- sound/soc/tegra/tegra210_mvc.c | 8 +- sound/soc/tegra/tegra210_sfc.c | 4 +- 14 files changed, 179 insertions(+), 87 deletions(-)

2 1

Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - edited 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was edited from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DYNAMIC_PIPELINE + 2]", I get the error: ALSA lib confeval.c:263:(snd_config_evaluate_string) wrong expression '$[$DYNAMIC_PIPELINE + 2]' ALSA lib conf.c:5632:(snd_config_expand_custom) Expand error (walk): Invalid argument Failed to expand pre-processor definitions in input config Is it possible to add support for some non-alphanumeric characters in the variable names , esp '_'? I tried this change it seems to suffice: ``` diff --git a/src/confeval.c b/src/confeval.c index a971bf38..7330d67e 100644 --- a/src/confeval.c +++ b/src/confeval.c @@ -190,7 +190,7 @@ int _snd_eval_string(snd_config_t **dst, const char *s, } else { e = s + 1; while (*e) { - if (!isalnum(*e)) + if (!isalnum(*e) && (*e != '_')) break; e++; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - edited 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was edited from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DYNAMIC_PIPELINE + 2]", I get the error: ALSA lib confeval.c:263:(snd_config_evaluate_string) wrong expression '$[$DYNAMIC_PIPELINE + 77]' ALSA lib conf.c:5632:(snd_config_expand_custom) Expand error (walk): Invalid argument Failed to expand pre-processor definitions in input config Is it possible to add support for some non-alphanumeric characters in the variable names , esp '_'? I tried this change it seems to suffice: ``` diff --git a/src/confeval.c b/src/confeval.c index a971bf38..7330d67e 100644 --- a/src/confeval.c +++ b/src/confeval.c @@ -190,7 +190,7 @@ int _snd_eval_string(snd_config_t **dst, const char *s, } else { e = s + 1; while (*e) { - if (!isalnum(*e)) + if (!isalnum(*e) && (*e != '_')) break; e++; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - opened 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was opened from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DMIC_CH_COUNT + 2]", I get the error: Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Issues with plughw working, however raw hw not on MAX98360A
by Dylanger Daly 07 Dec '21

07 Dec '21

Hi all, I'm currently having issues with my speakers on a Chromebook running the MAX98360A DAC. director@DRIFTWOOD:~$ aplay -D hw:sc7180rt5682max,1 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav XDG_RUNTIME_DIR (/run/user/0) is not owned by us (uid 1000), but by uid 0! (This could e.g. happen if you try to connect to a non-root PulseAudio as a root u) Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "hw:sc7180rt5682max,1": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 64 CHANNELS: 4 RATE: 48000 PERIOD_TIME: 64000 PERIOD_SIZE: 3072 PERIOD_BYTES: 24576 PERIODS: 2 BUFFER_TIME: 128000 BUFFER_SIZE: 6144 BUFFER_BYTES: 49152 TICK_TIME: ALL -------------------- aplay: set_params:1358: Channels count non available director@DRIFTWOOD:~$ aplay -D plughw:sc7180rt5682max,1 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav XDG_RUNTIME_DIR (/run/user/0) is not owned by us (uid 1000), but by uid 0! (This could e.g. happen if you try to connect to a non-root PulseAudio as a root u) Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "plughw:sc7180rt5682max,1": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED MMAP_COMPLEX RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE FLOAT64_LE FLOAT64_BE MU_LAW A_LAW IMA_AE SUBFORMAT: STD SAMPLE_BITS: [4 64] FRAME_BITS: [4 640000] CHANNELS: [1 10000] RATE: [4000 4294967295) PERIOD_TIME: 64000 PERIOD_SIZE: [256 274877907) PERIOD_BYTES: [128 4294967295) PERIODS: (0 2147484) BUFFER_TIME: [1 4294967295] BUFFER_SIZE: [512 549755813] BUFFER_BYTES: [256 4294967295] TICK_TIME: ALL -------------------- When I specify plughw, meaning "add software processing", I can hear perfect output, however if I attempt to use the raw hw handle, I get significantly less format support and the channel count isn't even known, and there is no sound output. I believe this is happening because the driver isn't being picked up, and plughw simply adds what it can to the device making it work. The end result is neither PipeWire or PluseAudio detect the speaker device, I assume this is because they're expecting to use the raw hw device handle Here's my ucm2 config: https://github.com/hexdump0815/imagebuilder/blob/d02a71ace7b93b8efbff842662… Any help would be appreciated, thank you

1 0

Support for 8086:ae20 Intel Smart Sound Technology on Fujitsu Lifebook
by Norbert Preining 07 Dec '21

07 Dec '21

Dear all, (please Cc) today I realized that the built-in microphone (next to the webcam) of my Fujitsu Lifebook U-series is not detected. (I usully use a headset, but due to travelling was forced to use the internal). I tried hard to get this device working, but neither with my self-compiled kernel (5.15.6) nor the Debian kernel distribution kernel 5.15 the microphone seems to be detected. I booted into Windows and tried to find out what the device might be. My **guess** is what is mentioned as Intel Smart Sound Technology with Vendor ID 8086 Device ID ae20 I git grepped the kernel sources for this, and snd-hda-intel contains quite some mentioning about Vendor ID 8086, but ae20 is not appearing. Now the interesting thing is that booting into linux and using lspci and lsusb, I cannot see a single device with the above specifications: lspci: root@sakefilet /usr/src/git/linux/Documentation # lspci 00:00.0 Host bridge: Intel Corporation Comet Lake-U v1 4c Host Bridge/DRAM Controller (rev 0c) 00:02.0 VGA compatible controller: Intel Corporation CometLake-U GT2 [UHD Graphics] (rev 02) 00:04.0 Signal processing controller: Intel Corporation Xeon E3-1200 v5/E3-1500 v5/6th Gen Core Processor Thermal Subsystem (rev 0c) 00:12.0 Signal processing controller: Intel Corporation Comet Lake Thermal Subsytem 00:14.0 USB controller: Intel Corporation Comet Lake PCH-LP USB 3.1 xHCI Host Controller 00:14.2 RAM memory: Intel Corporation Comet Lake PCH-LP Shared SRAM 00:14.3 Network controller: Intel Corporation Comet Lake PCH-LP CNVi WiFi 00:15.0 Serial bus controller [0c80]: Intel Corporation Serial IO I2C Host Controller 00:15.1 Serial bus controller [0c80]: Intel Corporation Comet Lake Serial IO I2C Host Controller 00:15.3 Serial bus controller [0c80]: Intel Corporation Device 02eb 00:16.0 Communication controller: Intel Corporation Comet Lake Management Engine Interface 00:16.3 Serial controller: Intel Corporation Device 02e3 00:1c.0 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #1 (rev f0) 00:1c.4 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #5 (rev f0) 00:1d.0 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #9 (rev f0) 00:1d.4 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #13 (rev f0) 00:1e.0 Communication controller: Intel Corporation Device 02a8 00:1e.1 Communication controller: Intel Corporation Device 02a9 00:1f.0 ISA bridge: Intel Corporation Comet Lake PCH-LP LPC Premium Controller/eSPI Controller 00:1f.3 Multimedia audio controller: Intel Corporation Comet Lake PCH-LP cAVS 00:1f.4 SMBus: Intel Corporation Comet Lake PCH-LP SMBus Host Controller 00:1f.5 Serial bus controller [0c80]: Intel Corporation Comet Lake SPI (flash) Controller 00:1f.6 Ethernet controller: Intel Corporation Ethernet Connection (10) I219-LM 06:00.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:00.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:01.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:02.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:04.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 08:00.0 System peripheral: Intel Corporation JHL7540 Thunderbolt 3 NHI [Titan Ridge 4C 2018] (rev 06) 3d:00.0 USB controller: Intel Corporation JHL7540 Thunderbolt 3 USB Controller [Titan Ridge 4C 2018] (rev 06) 72:00.0 Non-Volatile memory controller: Samsung Electronics Co Ltd NVMe SSD Controller SM981/PM981/PM983 77:00.0 SD Host controller: O2 Micro, Inc. SD/MMC Card Reader Controller (rev 01) lsusb: Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 004: ID 04f2:b6e1 Chicony Electronics Co., Ltd FJ Camera Bus 001 Device 003: ID 298d:2033 Next Biometrics NB-2033-U Bus 001 Device 002: ID 058f:9540 Alcor Micro Corp. AU9540 Smartcard Reader Bus 001 Device 005: ID 8087:0026 Intel Corp. AX201 Bluetooth Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub I attach the output of alsa-info for completeness. The kernel version is 5.15.6 with some patches from xannmod, but I could reproduce the same behaviour with standard Debian kernel 5.15. Am I missing some config option, or is the vendorid/devid not supported by now? Thanks for every suggestion, (Please CC!) Al the best Norbert -- PREINING Norbert https://www.preining.info Fujitsu Research + IFMGA Guide + TU Wien + TeX Live + Debian Dev GPG: 0x860CDC13 fp: F7D8 A928 26E3 16A1 9FA0 ACF0 6CAC A448 860C DC13

2 4

pre-processor only works with relative path based on input file with the complete path
by GitHub issues - opened 06 Dec '21

06 Dec '21

alsa-project/alsa-utils issue #123 was opened from ranj063: For ex: `alsatplg -P sof-hda-generic.conf -o test.conf` fails with: Unable to open included conf file sof-hda-generic.conf/include/components/eqiir/highpass_40hz_0db_48khz.conf Unable to process include file Failed to process includes Failed to process conditional includes in input config whereas: `alsatplg -P ~/work/sof/sof/tools/topology/topology2/sof-hda-generic.conf -o test.conf` works OK. Would it be possible to make the first command work with the use of the ALSA_CONFIG_DIR environment variable? Issue URL : https://github.com/alsa-project/alsa-utils/issues/123 Repository URL: https://github.com/alsa-project/alsa-utils

1 0