October 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: codecs: ES8326: Modify the configuration of and micbias
by Zhang Yi 28 Oct '24

28 Oct '24

Because we designed a new version of ES8326, the configuration of micbias needed to be modified.We tested the new driver, on both the new version and the old one. It works well. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..aa3e364827c8 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -614,6 +614,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 0x0F, 0x0F); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x00); + } } } else { if (!es8326->calibrated) { @@ -640,6 +644,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) ES8326_MUTE_MASK, ~(ES8326_MUTE)); } else { msleep(300); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); + } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 0x0F, 0x00); } @@ -821,7 +829,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) iface = snd_soc_component_read(comp, ES8326_HPDET_STA); dev_dbg(comp->dev, "gpio flag %#04x", iface); - if ((es8326->jack_remove_retry == 1) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 1) && (es8326->version < ES8326_VERSION_B)) { if (iface & ES8326_HPINSERT_FLAG) es8326->jack_remove_retry = 2; else @@ -859,7 +867,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ - if ((es8326->jack_remove_retry == 0) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 0) && (es8326->version < ES8326_VERSION_B)) { es8326->jack_remove_retry = 1; dev_dbg(comp->dev, "remove event check, invert HPJACK_POL, cnt = %d\n", es8326->jack_remove_retry); @@ -954,7 +962,7 @@ static int es8326_calibrate(struct snd_soc_component *component) regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); es8326->version = reg; - if ((es8326->version == ES8326_VERSION_B) && (es8326->calibrated == false)) { + if ((es8326->version >= ES8326_VERSION_B) && (es8326->calibrated == false)) { dev_dbg(component->dev, "ES8326_VERSION_B, calibrating\n"); regmap_write(es8326->regmap, ES8326_CLK_INV, 0xc0); regmap_write(es8326->regmap, ES8326_CLK_DIV1, 0x03); @@ -1047,7 +1055,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15); regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x80 | - ((es8326->version == ES8326_VERSION_B) ? + ((es8326->version >= ES8326_VERSION_B) ? (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol) : (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol | 0x04))); usleep_range(5000, 10000); @@ -1073,6 +1081,10 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x03); + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + } msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13
by Piyush Raj Chouhan 28 Oct '24

28 Oct '24

From: pc1598 <piyushchouhan1598(a)gmail.com> Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..075935707f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7551,6 +7551,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7996,6 +7997,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10989,6 +11000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.46.2

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[tiwai-sound:topic/compress-accel 1/1] sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used
by kernel test robot 25 Oct '24

25 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/compress-accel head: 04177158cf98a79744937893b100020d77e6f9ac commit: 04177158cf98a79744937893b100020d77e6f9ac [1/1] ALSA: compress_offload: introduce accel operation mode config: arm-randconfig-001-20241026 (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-lkp@…) compiler: arm-linux-gnueabi-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410260257.PNGCF9yq-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/compress_offload.c: In function 'snd_compr_poll': >> sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used [-Wunused-but-set-variable] 409 | struct snd_compr_runtime *runtime; | ^~~~~~~ vim +/runtime +409 sound/core/compress_offload.c 404 405 static __poll_t snd_compr_poll(struct file *f, poll_table *wait) 406 { 407 struct snd_compr_file *data = f->private_data; 408 struct snd_compr_stream *stream; > 409 struct snd_compr_runtime *runtime; 410 size_t avail; 411 __poll_t retval = 0; 412 413 if (snd_BUG_ON(!data)) 414 return EPOLLERR; 415 416 stream = &data->stream; 417 runtime = stream->runtime; 418 419 guard(mutex)(&stream->device->lock); 420 421 switch (stream->runtime->state) { 422 case SNDRV_PCM_STATE_OPEN: 423 case SNDRV_PCM_STATE_XRUN: 424 return snd_compr_get_poll(stream) | EPOLLERR; 425 default: 426 break; 427 } 428 429 poll_wait(f, &stream->runtime->sleep, wait); 430 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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Help to solve pop when sound loops
by Charles Mulder 25 Oct '24

25 Oct '24

Hi. I hope this is the correct place to ask for help relating to ALSA's C lib? I'm working on a wavetable project for college. Please see attached. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on buffer > period > frames, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C -- Sent with Tuta; enjoy secure & ad-free emails: https://tuta.com

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Help to solve pop when sound loops
by GitHub issues - edited 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was edited from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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Help to address pop at end of buffer
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was opened from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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ALC225 the microphone does not work in jack3.5
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #416 was opened from PavelGu: I have notebook clevo NL55AU with Ubuntu24.04 installed. The microphone does not work on a headset connected to a combined audio output jack 3.5. Logs: https://alsa-project.org/db/?f=89d2b631cb1635dd48357610d54e062bddef81b4 Can you help me? Issue URL : https://github.com/alsa-project/alsa-lib/issues/416 Repository URL: https://github.com/alsa-project/alsa-lib

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[Discussion] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #415 was opened from xeroxslayer: I raised this discussion a few months ago on the ALSA mailing list, but there was no response regarding this. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). Now, this profile will be autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software, I wouldn't know, I haven't actually tried adding a mono profile manually), but it seemed odd that one has to go through such trouble just to listen to mono sound... something that should be available out of the box IMO. Issue URL : https://github.com/alsa-project/alsa-lib/issues/415 Repository URL: https://github.com/alsa-project/alsa-lib

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Limit internal Mic boost on Dell platform
by Kailang 25 Oct '24

25 Oct '24

Hi Takashi, Attach patch will limit internal Mic boost on all Dell platform. BR, Kailang

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Update ucm conf for MediaTek Genio platforms
by GitHub pull_request - opened 24 Oct '24

24 Oct '24

alsa-project/alsa-ucm-conf pull request #456 was opened from zoran055: 1. Add dynamic configuration to detect HDMI/DP devices for mt8370-evk, mt8390-evk and mt8395-evk. 2. Add headset jack detection support for mt8370-evk, mt8390-evk and mt8395-evk. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/456 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/456.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.12-rc4
by Mark Brown 24 Oct '24

24 Oct '24

The following changes since commit 8cf0b93919e13d1e8d4466eb4080a4c4d9d66d7b: Linux 6.12-rc2 (2024-10-06 15:32:27 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc4 for you to fetch changes up to db7e59e6a39a4d3d54ca8197c796557e6d480b0d: ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc (2024-10-22 21:00:38 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.12 An uncomfortably large set of fixes due to me not getting round to sending them for longer than I should due to travel and illness. This is mostly smaller driver specific changes, but there are a couple of generic changes: - Bumping the minimal topology ABI we check for during validation, the code had support for v4 removed previously but the update of the define used for initial validation was missed. - Fix the assumption that DAPM structs will be embedded in a component which isn't true for card widgets when doing name comparisons, though fortunately this is rarely triggered. We've pulled in one Soundwire fix which was part of a larger series fixing cleanup issues in on Intel Soundwire systems. ---------------------------------------------------------------- Alexey Klimov (2): ASoC: qcom: sdm845: add missing soundwire runtime stream alloc ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs Amadeusz Sławiński (2): ASoC: Intel: avs: Update stream status in a separate thread ASoC: topology: Bump minimal topology ABI version Andrei Simion (1): MAINTAINERS: Update maintainer list for MICROCHIP ASOC, SSC and MCP16502 drivers Benjamin Bara (1): ASoC: dapm: avoid container_of() to get component Binbin Zhou (1): ASoC: loongson: Fix component check failed on FDT systems Chancel Liu (1): ASoC: fsl_micfil: Add a flag to distinguish with different volume control types Christian Heusel (1): ASoC: amd: yc: Add quirk for ASUS Vivobook S15 M3502RA Colin Ian King (1): ASoC: max98388: Fix missing increment of variable slot_found Derek Fang (1): ASoC: Intel: soc-acpi: lnl: Add match entry for TM2 laptops Ilya Dudikov (1): ASoC: amd: yc: Fix non-functional mic on ASUS E1404FA Jack Yu (1): ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue Kai Vehmanen (1): ASoC: SOF: Intel: hda-loader: do not wait for HDaudio IOC Kirill Marinushkin (1): ASoC: Change my e-mail to gmail Krzysztof Kozlowski (2): ASoC: qcom: Select missing common Soundwire module code on SDM845 ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc Lad Prabhakar (1): ASoC: rsnd: Fix probe failure on HiHope boards due to endpoint parsing Mark Brown (1): ASoC/SoundWire: clean up link DMA during stop for IPC4 Miquel Raynal (1): ASoC: dt-bindings: davinci-mcasp: Fix interrupt properties Ranjani Sridharan (4): ASoC: SOF: ipc4-topology: Do not set ALH node_id for aggregated DAIs ASoC: SOF: Intel: hda: Handle prepare without close for non-HDA DAI's soundwire: intel_ace2x: Send PDI stream number during prepare ASoC: SOF: Intel: hda: Always clean up link DMA during stop Shengjiu Wang (2): ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler ASoC: fsl_micfil: Add sample rate constraint Venkata Prasad Potturu (2): ASoC: SOF: amd: Add error log for DSP firmware validation failure ASoC: SOF: amd: Fix for ACP SRAM addr for acp7.0 platform Zhu Jun (1): ASoC: codecs: Fix error handling in aw_dev_get_dsp_status function Zichen Xie (1): ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe() .../bindings/sound/davinci-mcasp-audio.yaml | 18 ++--- MAINTAINERS | 5 +- drivers/soundwire/intel_ace2x.c | 19 ++--- include/uapi/sound/asoc.h | 2 +- sound/soc/amd/yc/acp6x-mach.c | 14 ++++ sound/soc/codecs/aw88399.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 15 ++-- sound/soc/codecs/max98388.c | 1 + sound/soc/codecs/pcm3060-i2c.c | 4 +- sound/soc/codecs/pcm3060-spi.c | 4 +- sound/soc/codecs/pcm3060.c | 4 +- sound/soc/codecs/pcm3060.h | 2 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/fsl/fsl_esai.c | 4 +- sound/soc/fsl/fsl_micfil.c | 81 +++++++++++++++++++++- sound/soc/intel/avs/core.c | 3 +- sound/soc/intel/avs/pcm.c | 19 +++++ sound/soc/intel/avs/pcm.h | 16 +++++ sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 38 ++++++++++ sound/soc/loongson/loongson_card.c | 1 + sound/soc/qcom/Kconfig | 2 + sound/soc/qcom/lpass-cpu.c | 2 + sound/soc/qcom/sc7280.c | 10 ++- sound/soc/qcom/sdm845.c | 7 +- sound/soc/sh/rcar/core.c | 7 +- sound/soc/soc-dapm.c | 4 +- sound/soc/sof/amd/acp-loader.c | 5 +- sound/soc/sof/amd/acp.c | 4 +- sound/soc/sof/intel/hda-dai-ops.c | 23 +++--- sound/soc/sof/intel/hda-dai.c | 37 ++++++++-- sound/soc/sof/intel/hda-loader.c | 17 ----- sound/soc/sof/ipc4-topology.c | 15 +++- 32 files changed, 298 insertions(+), 89 deletions(-) create mode 100644 sound/soc/intel/avs/pcm.h

2 1

[PATCH v4 0/7] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 24 Oct '24

24 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Add it to this patch set. Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Jaroslav Kysela (1): ALSA: compress_offload: introduce accel operation mode Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device .../sound/designs/compress-accel.rst | 136 ++++ include/sound/compress_driver.h | 46 ++ include/uapi/sound/compress_offload.h | 63 +- include/uapi/sound/compress_params.h | 23 +- sound/core/Kconfig | 3 + sound/core/compress_offload.c | 351 ++++++++- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 ++ sound/soc/fsl/fsl_asrc_m2m.c | 727 ++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 ++++++- sound/soc/fsl/fsl_easrc.h | 4 + 14 files changed, 1851 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/designs/compress-accel.rst create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

4 10

Demo application for accel operation.
by GitHub pull_request - opened 24 Oct '24

24 Oct '24

alsa-project/tinycompress pull request #26 was opened from TE-N-ShengjiuWang: This 'caccel' applicaton is a demo to show how to call the new introduced accel interface. The kernel patch is: ALSA: compress_offload: introduce accel operation mode As there is conflict when using the API of alsa-lib, add alsa_wrap file to separate the header files: alsa/asoundlib.h sound/asound.h command for conversion to 48kHz: caccel -i in.wav -o out.wav -r 48000 Request URL : https://github.com/alsa-project/tinycompress/pull/26 Patch URL : https://github.com/alsa-project/tinycompress/pull/26.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH v5 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 23 Oct '24

23 Oct '24

Hi Mark I got kernel-test report that v4 had compile warning. This is v5 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v4 -> v5 - fixup sumsung odroid compile warning - rebased on latest asoc/for-6.13 v3 -> v4 - adjust latest amd - rebased on asoc/for-6.13 v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87y138vbv7.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o73mqj5b.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 11 +-- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 584 deletions(-) -- 2.43.0

2 14

[PATCH v4 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 23 Oct '24

23 Oct '24

Hi Mark 2 weeks past. This is v4 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v3 -> v4 - adjust latest amd - rebased on asoc/for-6.13 v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87y138vbv7.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

2 14

Update default depop procedure
by Kailang 23 Oct '24

23 Oct '24

Hi Takashi, Attach patch will solve issue which Headphone has a chance no sound. Many Thanks. BR, Kailang

2 1

ALC4080 onboard audio support on the MSI X870 Tomahawk motherboard
by GitHub issues - opened 23 Oct '24

23 Oct '24

alsa-project/alsa-ucm-conf issue #455 was opened from nfp0: @perexg As mentioned [here](https://github.com/alsa-project/alsa-ucm-conf/issues/447#issuecomment…, it seems the ALC4080 audio on the new MSI X870 Tomahawk motherboard is not supported. USB ID is `0db0:cd0e` I have tested adding `cd0e` on the `USB-Audio.conf` file like so: ``` Regex "USB((0414:a0(0e|1[0124]))|(0b05:(19(84|9[69])|1a(16|2[07]|5[23c]|97|f1)))|(0db0:(cd0e|005a|124b|151f|1feb|3130|36e7|4(19c|22d|240|88c)|62a4|6c[0c]9|70d3|7696|82c7|8af7|961e|a(073|228|47c|74b)|b202|d1d7|d6e7))|(26ce:0a0[68]))" ``` With this, the microphone now seems to be correctly mapped and works. But the problem is that it's volume is way too low. It is set to 100% in both KDE and alsamixer (it's the Line one). ![image](https://github.com/user-attachments/assets/034e8f45-9e17-4338-9c1b-fda0b56dae29) How can I help solve this problem? Should I open a separate issue? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/455 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

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[PATCH 0/2] ASoC: fsl_xcvr: enable some interrupts
by Shengjiu Wang 23 Oct '24

23 Oct '24

Enable interrupt of cmdc status update and the interrupts for wrong preamble received. Shengjiu Wang (2): ASoC: fsl_xcvr: enable interrupt of cmdc status update ASoC: fsl_xcvr: reset RX dpath after wrong preamble sound/soc/fsl/fsl_xcvr.c | 94 ++++++++++++++++++++++++++++++++++------ sound/soc/fsl/fsl_xcvr.h | 5 +++ 2 files changed, 86 insertions(+), 13 deletions(-) -- 2.34.1

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ALSA problem , audio doesn't work
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubutnu
by GitHub issues - opened 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was opened from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue
by Jack Yu 22 Oct '24

22 Oct '24

clk_stop_timeout should be increased to 900ms to fix clock stop issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 87354bb1564e..d5c985ff5ac5 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -253,7 +253,7 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 200; + prop->clk_stop_timeout = 900; /* wake-up event */ prop->wake_capable = 1; -- 2.34.1

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