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[PATCH] ALSA: hda/proc - dump the coeff with the capital letters
by Hui Wang 09 Jul '21

09 Jul '21

Sometimes we need to dump the coeff to debug the audio issues, and need to compare it with the log of RTHDDump.exe under Windows, this is the coeff under Windows and Linux: Vendor widget coefficient value under Windows Index 0x00 0x0002 Index 0x01 0xAAAA Index 0x02 0x8AAA Processing caps: benign=0, ncoeff=91 under Linux Coeff 0x00: 0x0002 Coeff 0x01: 0xaaaa Coeff 0x02: 0x8aaa Windows prints the hex number with capital letter while Linux prints the hex number with lower case letter, it adds the trouble when comparing them. Let us change it to use capital letter. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 00c2eeb2c472..9037c87665b6 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -590,7 +590,7 @@ static void print_proc_caps(struct snd_info_buffer *buffer, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, i); val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0); - snd_iprintf(buffer, " Coeff 0x%02x: 0x%04x\n", i, val); + snd_iprintf(buffer, " Coeff 0x%02X: 0x%04X\n", i, val); } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, oldindex); } -- 2.25.1

2 2

[Patch v2] ALSA: compress: allow to leave draining state when pausing in draining
by Robert Lee 09 Jul '21

09 Jul '21

When compress offload pauses in draining state, not all platforms need to keep in draining state. Some platforms may call drain or partial drain again when resume from pause in draining, so it needs to wake up from snd_compress_wait_for_drain() in this case. Call API snd_compr_leave_draining_in_pause(), if the platform doesn't need to keep in draining state when pause in draining state. Signed-off-by: Robert Lee <lerobert(a)google.com> --- include/sound/compress_driver.h | 14 ++++++++++++++ sound/core/compress_offload.c | 8 +++++++- 2 files changed, 21 insertions(+), 1 deletion(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 277087f635f3..e16524a93a14 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -145,6 +145,7 @@ struct snd_compr_ops { * @lock: device lock * @device: device id * @use_pause_in_draining: allow pause in draining, true when set + * @leave_draining_in_pause: leave draining state when pausing in draining */ struct snd_compr { const char *name; @@ -156,6 +157,7 @@ struct snd_compr { struct mutex lock; int device; bool use_pause_in_draining; + bool leave_draining_in_pause; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -182,6 +184,18 @@ static inline void snd_compr_use_pause_in_draining(struct snd_compr_stream *subs substream->device->use_pause_in_draining = true; } +/** + * snd_compr_leave_draining_in_pause - Leave draining state when pause in draining + * @substream: compress substream to set + * + * In some platform, we need to leave draining state when we use pause in draining. + * Add API to allow leave draining state. + */ +static inline void snd_compr_leave_draining_in_pause(struct snd_compr_stream *substream) +{ + substream->device->leave_draining_in_pause = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 21ce4c056a92..c6e5c8f072d7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -719,8 +719,14 @@ static int snd_compr_pause(struct snd_compr_stream *stream) if (!stream->device->use_pause_in_draining) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) + if (!retval) { + if (stream->device->leave_draining_in_pause) { + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + wake_up(&stream->runtime->sleep); + break; + } stream->pause_in_draining = true; + } break; default: return -EPERM; -- 2.32.0.93.g670b81a890-goog

3 5

[PATCH v4] ASoC: soc-pcm: add a flag to reverse the stop sequence
by Vijendar Mukunda 09 Jul '21

09 Jul '21

From: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> On stream stop, currently CPU DAI stop sequence invoked first followed by DMA. For Few platforms, it is required to stop the DMA first before stopping CPU DAI. For Stoneyridge platform, it is required to invoke DMA driver stop first rather than invoking DWC I2S controller stop. Introduced new flag in dai_link structure for reordering stop sequence. Based on flag check, ASoC core will re-order the stop sequence. Fixes: 4378f1fbe92405 ("ASoC: soc-pcm: Use different sequence for start/stop trigger") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v3 -> v4: Updated Fixes tag in commit message v2 -> v3: moved "stop_dma_first" flag from card structure to dai_link structure and modified code to use. "stop_dma_first" flag in acp machine driver and soc pcm trigger api v1 -> v2: renamed flag as "stop_dma_fist" fixed build error by removing extra + symbol sound/soc/soc-pcm.c:1019:3: error: expected expression before 'struct' 1019 | + struct snd_soc_card *card = rtd->card; include/sound/soc.h | 6 ++++++ sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ sound/soc/soc-pcm.c | 22 ++++++++++++++++------ 3 files changed, 27 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 675849d07284..8e6dd8a257c5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -712,6 +712,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* This flag will reorder stop sequence. By enabling this flag + * DMA controller stop sequence will be invoked first followed by + * CPU DAI driver stop sequence + */ + unsigned int stop_dma_first:1; + #ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; /* For topology */ #endif diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 84e3906abd4f..9449fb40a956 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_rt5682_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_play_ops, SND_SOC_DAILINK_REG(designware1, rt5682, platform), }, @@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, SND_SOC_DAILINK_REG(designware2, rt5682, platform), }, @@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 46513bb97904..d1c570ca21ea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1015,6 +1015,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = -EINVAL, _ret = 0; int rollback = 0; @@ -1055,14 +1056,23 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); - if (ret < 0) - break; + if (rtd->dai_link->stop_dma_first) { + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; - ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); - if (ret < 0) - break; + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } else { + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } ret = snd_soc_link_trigger(substream, cmd, rollback); break; } -- 2.17.1

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[PATCH v4] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Randy Dunlap 08 Jul '21

08 Jul '21

On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Fixes: eb17726b00b3 ("m32r: add simple dma") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Acked-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- v2: rebase & resend; drop Cc: to Bo Shen <voice.shen(a)atmel.com> # bounced v3: drop "depends on HAS_DMA" (as suggested by Codrin; Thanks) v4: update Fixes: tag as suggested by Codrin (Thanks) --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) --- linux-next-20210707.orig/sound/soc/atmel/Kconfig +++ linux-next-20210707/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool

1 0

Can't get snd-aloop in sync with another device
by Mark Hills 08 Jul '21

08 Jul '21

I use dmix+snd-aloop for all local applications; then forwarded to the dmix+Layla3G using a custom C program. Despite asking for clock sync, they slowly drift. Am I doing something wrong? $ echo hw:Layla3G,0,0 > /proc/asound/Loopback/timer_source The forwarder drains its buffer and eventually underruns. Loopback is not providing enough audio to feed Layla3G -- drifts by about 9 samples/second at 48kHz. This _feels_ like a regression; I'm pretty sure this was perfectly in time before. I'd need to bisect kernel and alsa-lib changes. Below shows the status during playback/capture. The timer is definitely connected in some way. It's listed below, and aloop ticking corresponds to whether Layla3G is open by any application (not just the forwarder) -- Mark $ cat /proc/asound/Loopback/cable#0 Cable 0 substream 0: inactive Cable 0 substream 1: inactive Cable 0 substream 2: inactive Cable 0 substream 3: valid: 3 running: 3 pause: 0 Playback buffer_size: 65536 buffer_pos: 12800 silent_size: 0 period_size: 512 bytes_per_sec: 384000 sample_align: 8 rate_shift: 0 sound timer: hw:0,0,0 timer open: playback Capture buffer_size: 65536 buffer_pos: 53760 silent_size: 0 period_size: 512 bytes_per_sec: 384000 sample_align: 8 rate_shift: 0 sound timer: hw:0,0,0 timer open: playback Cable 0 substream 4: inactive Cable 0 substream 5: inactive Cable 0 substream 6: inactive Cable 0 substream 7: inactive

1 0

trigger timestamp
by Muni Sekhar 08 Jul '21

08 Jul '21

Hi all, >From the user space test application , Is it possible to get the trigger timestamp corresponding to SNDRV_PCM_TRIGGER_START & SNDRV_PCM_TRIGGER_STOP events for audio playback capture? -- Thanks, Sekhar

2 5

Patch For Lenovo Legion 7 15IMHg05 & Yoga 7 14ITL5
by Cameron Berkenpas 08 Jul '21

08 Jul '21

Hello, I have a patch against 5.12.x that results in working speaker audio for the Lenovo Legion 7 15IMHg05 and the Lenovo Yoga 7 14ITL5. How should I go about getting this merged? Should this be against the latest torvalds git? Is this the right place to present this particular patch? There are no issues checking my changes against checkpatch.pl. Thanks! diff --git a/sound/pci/hda/legion_15imhg05_helper.c b/sound/pci/hda/legion_15imhg05_helper.c new file mode 100644 index 000000000000..6ead999ec3c9 --- /dev/null +++ b/sound/pci/hda/legion_15imhg05_helper.c @@ -0,0 +1,41 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Legion 15IMHg05 speaker output. */ + +static const struct hda_verb legion_15imhg05_coefs[] = { +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x41 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, + +// Set right speaker for Legion 7i. +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0xc }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2a }, +{0x20, 0x4b0, 0x20 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, +{} +}; + +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_INIT: + spec->gen.automute_speaker = 0; + break; + } +} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e46e43dac6bf..28a4955cf2c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6352,6 +6352,12 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" +/* For alc287_fixup_legion_15imhg05_speakers() */ +#include "legion_15imhg05_helper.c" + +/* For yoga7_14itl5_coefs */ +#include "yoga7_14itl5_coefs.c" + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6561,6 +6567,9 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, }; static const struct hda_fixup alc269_fixups[] = { @@ -8088,6 +8097,24 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = legion_15imhg05_coefs, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = yoga7_14itl5_coefs, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, [ALC295_FIXUP_ASUS_DACS] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_asus_dacs, @@ -8520,6 +8547,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Lenovo Legion 7", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_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), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), diff --git a/sound/pci/hda/yoga7_14itl5_coefs.c b/sound/pci/hda/yoga7_14itl5_coefs.c new file mode 100644 index 000000000000..a437fe042a1d --- /dev/null +++ b/sound/pci/hda/yoga7_14itl5_coefs.c @@ -0,0 +1,29 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Yoga 7 14ITL5 speaker output. */ + +static const struct hda_verb yoga7_14itl5_coefs[] = { +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x41 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, + +// set right speaker Yoga 7i. +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0xc }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2a }, +{0x20, 0x4b0, 0x20 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, +{} +}; +

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Audio out on DisplayPort but not HDMI, on AMD Ryzen APU/Vega
by Damjan Georgievski 08 Jul '21

08 Jul '21

I have a desktop computer with an AMD Ryzen 5 4650G PRO APU (CPU + on-die GPU). The motherboard (MSI Mortar B550m wifi / bios 1.6). has one displayport and one HDMI outputs. The displayport is connected to my Lenovo P27 monitor, and the HDMI to my Samsung TV. When I choose the GPU as an audio out, I only get audio on the 3.5" headphones jack of the Lenovo monitor, while I would like to get the audio on the Samsung TV over HDMI. I can't find a way to switch the different outputs (neither over pipewire or just using native alsa). pavucontrol configuration https://i.imgur.com/9nbVtNz.png alsamixer -c0 only has only one option to mute s/pdif, which mutes the output on the displayport -> Lenovo -> 3.5" jack. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Generic_1 [HD-Audio Generic], device 0: ALCS1200A Analog [ALCS1200A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Generic_1 [HD-Audio Generic], device 1: ALCS1200A Digital [ALCS1200A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 Am I correct that card 1 here ^ should have at least two devices? distro is Archlinux kernel is 5.12.13-arch1-2 alsa-* 1.2.5.1-3 pipewire 1:0.3.30-2 plasma 5.22.2.1-1 xorg-server 1.20.11-1 amdgpu vanilla driver $ aplay -L null Discard all samples (playback) or generate zero samples (capture) sysdefault Default Audio Device samplerate Rate Converter Plugin Using Samplerate Library speexrate Rate Converter Plugin Using Speex Resampler jack JACK Audio Connection Kit oss Open Sound System pulse PulseAudio Sound Server speex Plugin using Speex DSP (resample, agc, denoise, echo, dereverb) upmix Plugin for channel upmix (4,6,8) vdownmix Plugin for channel downmix (stereo) with a simple spacialization default Default ALSA Output (currently PulseAudio Sound Server) usbstream:CARD=C93 罗技高清网络摄像机 C93 USB Stream Output hdmi:CARD=Generic,DEV=0 HD-Audio Generic, HDMI 0 HDMI Audio Output usbstream:CARD=Generic HD-Audio Generic USB Stream Output sysdefault:CARD=Generic_1 HD-Audio Generic, ALCS1200A Analog Default Audio Device front:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog Front output / input surround21:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Digital IEC958 (S/PDIF) Digital Audio Output usbstream:CARD=Generic_1 HD-Audio Generic USB Stream Output $ lspci 00:00.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Root Complex 00:00.2 IOMMU: Advanced Micro Devices, Inc. [AMD] Renoir IOMMU 00:01.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe Dummy Host Bridge 00:02.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe Dummy Host Bridge 00:02.1 PCI bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe GPP Bridge 00:02.2 PCI bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe GPP Bridge 00:08.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe Dummy Host Bridge 00:08.1 PCI bridge: Advanced Micro Devices, Inc. [AMD] Renoir Internal PCIe GPP Bridge to Bus 00:14.0 SMBus: Advanced Micro Devices, Inc. [AMD] FCH SMBus Controller (rev 51) 00:14.3 ISA bridge: Advanced Micro Devices, Inc. [AMD] FCH LPC Bridge (rev 51) 00:18.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 0 00:18.1 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 1 00:18.2 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 2 00:18.3 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 3 00:18.4 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 4 00:18.5 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 5 00:18.6 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 6 00:18.7 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 7 16:00.0 USB controller: Advanced Micro Devices, Inc. [AMD] Device 43ee 16:00.1 SATA controller: Advanced Micro Devices, Inc. [AMD] Device 43eb 16:00.2 PCI bridge: Advanced Micro Devices, Inc. [AMD] Device 43e9 20:08.0 PCI bridge: Advanced Micro Devices, Inc. [AMD] Device 43ea 20:09.0 PCI bridge: Advanced Micro Devices, Inc. [AMD] Device 43ea 29:00.0 Network controller: Intel Corporation Wi-Fi 6 AX200 (rev 1a) 2a:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8125 2.5GbE Controller (rev 04) 2b:00.0 Non-Volatile memory controller: ADATA Technology Co., Ltd. XPG SX8200 Pro PCIe Gen3x4 M.2 2280 Solid State Drive (rev 03) 30:00.0 VGA compatible controller: Advanced Micro Devices, Inc. [AMD/ATI] Renoir (rev d9) 30:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device 1637 30:00.2 Encryption controller: Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) Platform Security Processor 30:00.3 USB controller: Advanced Micro Devices, Inc. [AMD] Renoir USB 3.1 30:00.4 USB controller: Advanced Micro Devices, Inc. [AMD] Renoir USB 3.1 30:00.6 Audio device: Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) HD Audio Controller $ cat /proc/asound/devices 1: : sequencer 2: [ 1- 3]: digital audio playback 3: [ 1- 0]: hardware dependent 4: [ 1] : control 5: [ 0- 0]: digital audio capture 6: [ 0] : control 7: [ 2- 0]: digital audio playback 8: [ 2- 0]: digital audio capture 9: [ 2- 1]: digital audio playback 10: [ 2- 2]: digital audio capture 11: [ 2- 0]: hardware dependent 12: [ 2] : control 33: : timer $ cat /proc/asound/cards 0 [C93 ]: USB-Audio - 罗技高清网络摄像机 C93 罗技高清网络摄像机 C93 at usb-0000:30:00.3-1, high speed 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfcb88000 irq 95 2 [Generic_1 ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfcb80000 irq 96 $ cat /proc/asound/card card0/ card1/ card2/ cards -- damjan

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ALSA: intel8x0: div by zero in snd_intel8x0_update()
by Sergey Senozhatsky 08 Jul '21

08 Jul '21

Hi, I'm running (sometimes) into the following problem during resume divide error: 0000 [#1] PREEMPT SMP NOPTI RIP: 0010:snd_intel8x0_interrupt+0x121/0x279 Code: 42 8b 44 35 34 41 0f af c5 42 03 44 35 38 42 89 44 35 38 48 8b 0c 24 80 b9 60 03 00 00 00 78 0f 49 8d 0c 2e 48 83 c1 38 31 d2 <f7> 71 f4 89 11 42 8b 7c 35 48 44 01 ef 83 e7 1f 42 89 7c 35 48 48 RSP: 0000:ffff9a0a80108eb0 EFLAGS: 00010046 RAX: 0000000000000000 RBX: 0000000000000019 RCX: ffff90d8c5efc198 RDX: 0000000000000000 RSI: ffff9a0a80549016 RDI: ffff9a0a80549024 RBP: ffff90d8c5efc060 R08: 000000000000197a R09: 00000f604ed00191 R10: 00000000000001e0 R11: ffffffff9468e1d8 R12: 0000000000000020 R13: 0000000000000040 R14: 0000000000000100 R15: 0000000000000002 FS: 00007a75c397aff8(0000) GS:ffff90d912d80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007a77945d1000 CR3: 000000015bf46002 CR4: 0000000000360ea0 Call Trace: <IRQ> __handle_irq_event_percpu+0xa0/0x1c0 handle_irq_event_percpu+0x2d/0x70 handle_irq_event+0x2c/0x48 handle_fasteoi_irq+0xa1/0x161 do_IRQ+0x51/0xd6 common_interrupt+0xf/0xf </IRQ> RIP: 0033:0x7a7856462c59 Code: 89 ca 48 2b 57 20 48 83 c2 10 31 c0 48 3b 57 28 48 0f 46 c1 c3 cc cc cc cc cc cc cc cc cc cc cc cc 64 48 8b 0c 25 00 00 00 00 <b8> f8 02 00 00 48 03 41 08 c3 cc cc cc cc cc cc cc cc cc cc cc cc RSP: 002b:00007a75c39794e8 EFLAGS: 00000246 ORIG_RAX: ffffffffffffffde RAX: 02fa413b24209c6c RBX: 0000017f19e1cf9e RCX: 00007a75c397aff8 RDX: 00007a7855792472 RSI: 00007a7855790aa0 RDI: 0000000000000005 RBP: 0000000000000005 R08: 0000000000000012 R09: 000000000000000d R10: 00000000009f86d2 R11: 000000000000197a R12: 0000017f19e40e7d R13: 000005ee937ae557 R14: 00007a7855790aa0 R15: 00007a7855792472 Modules linked in: ---[ end trace 2ef6d63d0e3d757c ]--- RIP: 0010:snd_intel8x0_interrupt+0x121/0x279 Code: 42 8b 44 35 34 41 0f af c5 42 03 44 35 38 42 89 44 35 38 48 8b 0c 24 80 b9 60 03 00 00 00 78 0f 49 8d 0c 2e 48 83 c1 38 31 d2 <f7> 71 f4 89 11 42 8b 7c 35 48 44 01 ef 83 e7 1f 42 89 7c 35 48 48 RSP: 0000:ffff9a0a80108eb0 EFLAGS: 00010046 RAX: 0000000000000000 RBX: 0000000000000019 RCX: ffff90d8c5efc198 RDX: 0000000000000000 RSI: ffff9a0a80549016 RDI: ffff9a0a80549024 RBP: ffff90d8c5efc060 R08: 000000000000197a R09: 00000f604ed00191 R10: 00000000000001e0 R11: ffffffff9468e1d8 R12: 0000000000000020 R13: 0000000000000040 R14: 0000000000000100 R15: 0000000000000002 FS: 00007a75c397aff8(0000) GS:ffff90d912d80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007a77945d1000 CR3: 000000015bf46002 CR4: 0000000000360ea0 This corresponds to ichdev->position %= ichdev->size; in snd_intel8x0_update(). A print out of that ichdev looks as follows snd_intel8x0 0000:00:18.0: lvi_frag = 0, frags = 0, size = 0, period_size = 0x0, period_size1 = 0x0

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[PATCH] ALSA: intel8x0: Fix breakage at ac97 clock measurement
by Takashi Iwai 08 Jul '21

08 Jul '21

The recent workaround for the wild interrupts in commit c1f0616124c4 ("ALSA: intel8x0: Don't update period unless prepared") leaded to a regression, causing the interrupt storm during ac97 clock measurement at the driver probe. We need to handle the interrupt while the clock measurement as well as the proper PCM streams. Fixes: c1f0616124c4 ("ALSA: intel8x0: Don't update period unless prepared") Reported-and-tested-by: Max Filippov <jcmvbkbc(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CAMo8BfKKMQkcsbOQaeEjq_FsJhdK=fn598dvh7YOcZshUSOH… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/intel8x0.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 2d1bfbcba933..df3ba5c70de9 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -694,7 +694,7 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich int status, civ, i, step; int ack = 0; - if (!ichdev->prepared || ichdev->suspended) + if (!(ichdev->prepared || chip->in_measurement) || ichdev->suspended) return; spin_lock_irqsave(&chip->reg_lock, flags); -- 2.26.2

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