April 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/3] ASoC: cleanup mutex lock
by Kuninori Morimoto 17 Apr '23

17 Apr '23

Hi Mark ASoC is using many type of mutex lock, but some of them has helper function, but some doesn't. Or, it has helper function, but is static. This patch-set adds helper function and use it. Kuninori Morimoto (3): ASoC: expand snd_soc_dapm_mutex_lock/unlock() ASoC: expand snd_soc_dpcm_mutex_lock/unlock() ASoC: add snd_soc_card_mutex_lock/unlock() include/sound/soc-card.h | 17 +++++- include/sound/soc-dapm.h | 5 -- include/sound/soc.h | 105 +++++++++++++++++++++++++++++++-- sound/soc/soc-component.c | 12 ++-- sound/soc/soc-compress.c | 60 +++++++++---------- sound/soc/soc-core.c | 8 +-- sound/soc/soc-dapm.c | 119 +++++++++++++++++++------------------- sound/soc/soc-pcm.c | 17 +----- 8 files changed, 216 insertions(+), 127 deletions(-) -- 2.25.1

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[PATCH 2/2] ASoC: fsl_mqs: call pm_runtime_disable() on error path
by Liliang Ye 17 Apr '23

17 Apr '23

pm_runtime_disable was missed in cleanup operation, which corresponds to the earlier call to pm_runtime_enable. To fix this, add pm_runtime_disable() on error path. Fixes: 9e28f6532c61 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by: Liliang Ye <yll(a)hust.edu.cn> Reviewed-by: Dan Carpenter <error27(a)gmail.com> --- sound/soc/fsl/fsl_mqs.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index 32d20d351bbf..129d426c60c4 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -251,9 +251,13 @@ static int fsl_mqs_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, &fsl_mqs_dai, 1); if (ret) - return ret; + goto err_pm_disable; return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); + return ret; } static int fsl_mqs_remove(struct platform_device *pdev) -- 2.34.1

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[PATCH v2 0/7] soundwire: qcom: add support for SM8550 (Soundwire v2.0.0)
by Krzysztof Kozlowski 17 Apr '23

17 Apr '23

Hi, Changes since v1: 1. Patch 1: Increase maxItems to 16 for port-related properties. 2. Re-order patch 1 and 2. 3. Patch 3: Drop unneeded semicolon. 4. Patch 5: Fix lang typo in subject. Best regards, Krzysztof Krzysztof Kozlowski (7): dt-bindings: soundwire: qcom: add Qualcomm Soundwire v2.0.0 dt-bindings: soundwire: qcom: add 16-bit sample interval soundwire: qcom: allow 16-bit sample interval for ports soundwire: qcom: use consistently 'ctrl' as state variable name soundwire: qcom: prepare for handling different register layouts soundwire: qcom: add support for v2.0.0 controller soundwire: qcom: use tabs for indentation in defines .../bindings/soundwire/qcom,soundwire.yaml | 41 +- drivers/soundwire/qcom.c | 387 ++++++++++++------ 2 files changed, 289 insertions(+), 139 deletions(-) -- 2.34.1

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[PATCH resend 0/3] ASoC: cleanup mutex lock
by Kuninori Morimoto 17 Apr '23

17 Apr '23

Hi Mark This is resend patch-set. ASoC is using many type of mutex lock, but some of them has helper function, but some doesn't. Or, it has, but can't be used because of static. This patch-set adds helper function and use it. Kuninori Morimoto (3): ASoC: expand snd_soc_dapm_mutex_lock/unlock() ASoC: expand snd_soc_dpcm_mutex_lock/unlock() ASoC: add snd_soc_card_mutex_lock/unlock() include/sound/soc-card.h | 17 +++++- include/sound/soc-dapm.h | 5 -- include/sound/soc.h | 105 +++++++++++++++++++++++++++++++-- sound/soc/soc-component.c | 12 ++-- sound/soc/soc-compress.c | 60 +++++++++---------- sound/soc/soc-core.c | 8 +-- sound/soc/soc-dapm.c | 119 +++++++++++++++++++------------------- sound/soc/soc-pcm.c | 17 +----- 8 files changed, 216 insertions(+), 127 deletions(-) -- 2.25.1

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USB-Audio: ALC4080 on Gigabyte Z690 AORUS ULTRA
by GitHub pull_request - opened 16 Apr '23

16 Apr '23

alsa-project/alsa-ucm-conf pull request #304 was opened from MichailiK: This PR adds the USB ID of the ALC4080 found in the [Z690 AORUS ULTRA](https://www.gigabyte.com/Motherboard/Z690-AORUS-ULTRA-rev-1x/sp#sp), `0414:a011`. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/304 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/304.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALSA: arecord : silennce recorded as 0x80
by Muni Sekhar 15 Apr '23

15 Apr '23

Hi all, I recorded the silence on the default audio card by running "arecord -d 5 test.wav." Once it exited, I checked the contents by running "hexdump test.wav," and I see that silence is recorded as a sequence of 0x80 instead of 0x00. 0000000 4952 4646 0024 8000 4157 4556 6d66 2074 0000010 0010 0000 0001 0001 1f40 0000 1f40 0000 0000020 0001 0008 6164 6174 0000 8000 8080 8080 0000030 8080 8080 8080 8080 8080 8080 8080 8080 0000040 8080 8080 8080 8080 8080 8080 8080 8080 ... What might be the reason for silence being recorded as 0x80 instead of 0x0? -- Thanks, Sekhar

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arecord: duration incorrect when ALSA chooses different sample rate
by GitHub pull_request - opened 15 Apr '23

15 Apr '23

alsa-project/alsa-utils pull request #200 was opened from iam-TJ: I encountered this issue whilst researching a bug in capture handling in ALSA/PulseAudio/PortAudio/Audacity. When recording for a fixed duration the duration is measured by the quantity of samples, not clock time. In `capture()` a call to `calc_count()` occurs before `set_params()`, but in that latter function the actual sample rate used may differ from the requested rate , but the count is not recalculated. E.g: ``` $ time arecord -D hw:1,0 -d 2 -c 2 -f S16_LE -r 384000 --dump-hw-params /tmp/test.11.wav Recording WAVE '/tmp/test.11.wav' : Signed 16 bit Little Endian, Rate 384000 Hz, Stereo HW Params of device "hw:1,0": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S32_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [32 64] CHANNELS: 2 RATE: [44100 192000] PERIOD_TIME: (83 11888617) PERIOD_SIZE: [16 524288] PERIOD_BYTES: [128 2097152] PERIODS: [2 32] BUFFER_TIME: (166 23777234) BUFFER_SIZE: [32 1048576] BUFFER_BYTES: [128 4194304] TICK_TIME: ALL -------------------- Warning: rate is not accurate (requested = 384000Hz, got = 192000Hz) please, try the plug plugin real 0m4.009s user 0m0.004s sys 0m0.007s ``` In this example, because the actual sample rate was 1/2 of that requested, count is double what it should be and therefore the recording continues for 4 seconds instead of the requested 2 seconds. Moving the call to `set_params()` to before the count logic solves it: ``` $ time aplay/aplay -C -D hw:1,0 -d 2 -c 2 -f S16_LE -r 384000 --dump-hw-params /tmp/test.11.wav HW Params of device "hw:1,0": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S32_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [32 64] CHANNELS: 2 RATE: [44100 192000] PERIOD_TIME: (83 11888617) PERIOD_SIZE: [16 524288] PERIOD_BYTES: [128 2097152] PERIODS: [2 32] BUFFER_TIME: (166 23777234) BUFFER_SIZE: [32 1048576] BUFFER_BYTES: [128 4194304] TICK_TIME: ALL -------------------- Warning: rate is not accurate (requested = 384000Hz, got = 192000Hz) please, try the plug plugin Recording WAVE '/tmp/test.11.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Stereo real 0m2.009s user 0m0.001s sys 0m0.008s ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/200 Patch URL : https://github.com/alsa-project/alsa-utils/pull/200.patch Repository URL: https://github.com/alsa-project/alsa-utils

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ALSA: struct snd_pcm_hardware.formats
by Muni Sekhar 14 Apr '23

14 Apr '23

Hi all, struct snd_pcm_hardware.formats (SNDRV_PCM_FMTBIT_S8, SNDRV_PCM_FMTBIT_S16_LE, SNDRV_PCM_FMTBIT_U16_LE, SNDRV_PCM_FMTBIT_S16_BE, etc.), are these sound card hardware capabilities or something that needs to be taken care of on the software (driver or userspace) side? -- Thanks, Sekhar

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[PATCH 1/2] ALSA: pcm: rewrite snd_pcm_playback_silence()
by Oswald Buddenhagen 14 Apr '23

14 Apr '23

The auto-silencer supports two modes: "thresholded" to fill up "just enough", and "top-up" to fill up "as much as possible". The two modes used rather distinct code paths, which this patch unifies. The only remaining distinction is how much we actually want to fill. This fixes a bug in thresholded mode, where we failed to use new_hw_ptr, resulting in under-fill. Top-up mode is now more well-behaved and much easier to understand in corner cases. This also updates comments in the proximity of silencing-related data structures. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- .../kernel-api/writing-an-alsa-driver.rst | 14 +-- include/sound/pcm.h | 14 +-- include/uapi/sound/asound.h | 9 +- sound/core/pcm_lib.c | 86 ++++++++----------- sound/core/pcm_local.h | 3 +- sound/core/pcm_native.c | 6 +- 6 files changed, 62 insertions(+), 70 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index a368529e8ed3..f2834a016473 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -1577,14 +1577,16 @@ are the contents of this file: unsigned int period_step; unsigned int sleep_min; /* min ticks to sleep */ snd_pcm_uframes_t start_threshold; - snd_pcm_uframes_t stop_threshold; - snd_pcm_uframes_t silence_threshold; /* Silence filling happens when - noise is nearest than this */ - snd_pcm_uframes_t silence_size; /* Silence filling size */ + // The following two thresholds alleviate playback buffer underruns; when + // hw_avail drops below the threshold, the respective action is triggered: + snd_pcm_uframes_t stop_threshold; /* stop playback */ + snd_pcm_uframes_t silence_threshold; /* pre-fill buffer with silence */ + snd_pcm_uframes_t silence_size; /* msx size of silence pre-fill */ snd_pcm_uframes_t boundary; /* pointers wrap point */ - snd_pcm_uframes_t silenced_start; - snd_pcm_uframes_t silenced_size; + // internal data of auto-silencer + snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ + snd_pcm_uframes_t silence_filled; /* size filled with silence */ snd_pcm_sync_id_t sync; /* hardware synchronization ID */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 27040b472a4f..f20400bb7032 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -378,18 +378,18 @@ struct snd_pcm_runtime { unsigned int rate_den; unsigned int no_period_wakeup: 1; - /* -- SW params -- */ - int tstamp_mode; /* mmap timestamp is updated */ + /* -- SW params; see struct snd_pcm_sw_params for comments -- */ + int tstamp_mode; unsigned int period_step; snd_pcm_uframes_t start_threshold; snd_pcm_uframes_t stop_threshold; - snd_pcm_uframes_t silence_threshold; /* Silence filling happens when - noise is nearest than this */ - snd_pcm_uframes_t silence_size; /* Silence filling size */ - snd_pcm_uframes_t boundary; /* pointers wrap point */ + snd_pcm_uframes_t silence_threshold; + snd_pcm_uframes_t silence_size; + snd_pcm_uframes_t boundary; + // internal data of auto-silencer snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ - snd_pcm_uframes_t silence_filled; /* size filled with silence */ + snd_pcm_uframes_t silence_filled; /* already filled part of silence area */ union snd_pcm_sync_id sync; /* hardware synchronization ID */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index de6810e94abe..50aa1d98873f 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -429,9 +429,12 @@ struct snd_pcm_sw_params { snd_pcm_uframes_t avail_min; /* min avail frames for wakeup */ snd_pcm_uframes_t xfer_align; /* obsolete: xfer size need to be a multiple */ snd_pcm_uframes_t start_threshold; /* min hw_avail frames for automatic start */ - snd_pcm_uframes_t stop_threshold; /* min avail frames for automatic stop */ - snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ - snd_pcm_uframes_t silence_size; /* silence block size */ + // The following two thresholds alleviate playback buffer underruns; when + // hw_avail drops below the threshold, the respective action is triggered: + snd_pcm_uframes_t stop_threshold; /* stop playback */ + snd_pcm_uframes_t silence_threshold; /* pre-fill buffer with silence */ + snd_pcm_uframes_t silence_size; /* max size of silence pre-fill; when >= boundary, + * fill played area with silence immediately */ snd_pcm_uframes_t boundary; /* pointers wrap point */ unsigned int proto; /* protocol version */ unsigned int tstamp_type; /* timestamp type (req. proto >= 2.0.12) */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 02fd65993e7e..b074c5b926db 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -42,70 +42,58 @@ static int fill_silence_frames(struct snd_pcm_substream *substream, * * when runtime->silence_size >= runtime->boundary - fill processed area with silence immediately */ -void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr) +void snd_pcm_playback_silence(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); + snd_pcm_sframes_t hw_avail, added, noise_dist; snd_pcm_uframes_t frames, ofs, transfer; int err; + // This will "legitimately" turn negative on underrun, and will be mangled + // into a huge number by the boundary crossing handling. The initial state + // might also be not quite sane. The code below MUST account for these cases. + hw_avail = appl_ptr - runtime->status->hw_ptr; + if (hw_avail < 0) + hw_avail += runtime->boundary; + else if ((snd_pcm_uframes_t) hw_avail >= runtime->boundary) + hw_avail -= runtime->boundary; + + added = appl_ptr - runtime->silence_start; + if (added) { + if (added < 0) + added += runtime->boundary; + if ((snd_pcm_uframes_t)added < runtime->silence_filled) + runtime->silence_filled -= added; + else + runtime->silence_filled = 0; + runtime->silence_start = appl_ptr; + } + + noise_dist = hw_avail + runtime->silence_filled; if (runtime->silence_size < runtime->boundary) { - snd_pcm_sframes_t noise_dist, n; - snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); - if (runtime->silence_start != appl_ptr) { - n = appl_ptr - runtime->silence_start; - if (n < 0) - n += runtime->boundary; - if ((snd_pcm_uframes_t)n < runtime->silence_filled) - runtime->silence_filled -= n; - else - runtime->silence_filled = 0; - runtime->silence_start = appl_ptr; - } - if (runtime->silence_filled >= runtime->buffer_size) - return; - noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled; - if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold) - return; frames = runtime->silence_threshold - noise_dist; + if ((snd_pcm_sframes_t) frames <= 0) + return; if (frames > runtime->silence_size) frames = runtime->silence_size; } else { - if (new_hw_ptr == ULONG_MAX) { /* initialization */ - snd_pcm_sframes_t avail = snd_pcm_playback_hw_avail(runtime); - if (avail > runtime->buffer_size) - avail = runtime->buffer_size; - runtime->silence_filled = avail > 0 ? avail : 0; - runtime->silence_start = (runtime->status->hw_ptr + - runtime->silence_filled) % - runtime->boundary; - } else { - ofs = runtime->status->hw_ptr; - frames = new_hw_ptr - ofs; - if ((snd_pcm_sframes_t)frames < 0) - frames += runtime->boundary; - runtime->silence_filled -= frames; - if ((snd_pcm_sframes_t)runtime->silence_filled < 0) { - runtime->silence_filled = 0; - runtime->silence_start = new_hw_ptr; - } else { - runtime->silence_start = ofs; - } - } - frames = runtime->buffer_size - runtime->silence_filled; + frames = runtime->buffer_size - noise_dist; + if ((snd_pcm_sframes_t) frames <= 0) + return; } + if (snd_BUG_ON(frames > runtime->buffer_size)) return; - if (frames == 0) - return; - ofs = runtime->silence_start % runtime->buffer_size; - while (frames > 0) { + ofs = (runtime->silence_start + runtime->silence_filled) % runtime->buffer_size; + do { transfer = ofs + frames > runtime->buffer_size ? runtime->buffer_size - ofs : frames; err = fill_silence_frames(substream, ofs, transfer); snd_BUG_ON(err < 0); runtime->silence_filled += transfer; frames -= transfer; ofs = 0; - } + } while (frames > 0); snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); } @@ -439,10 +427,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return 0; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, new_hw_ptr); - if (in_interrupt) { delta = new_hw_ptr - runtime->hw_ptr_interrupt; if (delta < 0) @@ -460,6 +444,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_wrap += runtime->boundary; } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + runtime->silence_size > 0) + snd_pcm_playback_silence(substream); + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index ecb21697ae3a..42fe3a4e9154 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -29,8 +29,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); -void snd_pcm_playback_silence(struct snd_pcm_substream *substream, - snd_pcm_uframes_t new_hw_ptr); +void snd_pcm_playback_silence(struct snd_pcm_substream *substream); static inline snd_pcm_uframes_t snd_pcm_avail(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 331380c2438b..0e3e7997dc58 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -958,7 +958,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (snd_pcm_running(substream)) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); @@ -1455,7 +1455,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, __snd_pcm_set_state(runtime, state); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } @@ -1916,7 +1916,7 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *substream, runtime->control->appl_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); snd_pcm_stream_unlock_irq(substream); } -- 2.40.0.152.g15d061e6df

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[PATCH] ALSA: hda/hdmi: disable KAE for Intel DG2
by Kai Vehmanen 14 Apr '23

14 Apr '23

Use of keep-alive (KAE) has resulted in loss of audio on some A750/770 cards as the transition from keep-alive to stream playback is not working as expected. As there is limited benefit of the new KAE mode on discrete cards, revert back to older silent-stream implementation on these systems. Cc: stable(a)vger.kernel.org Fixes: 15175a4f2bbb ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2") Link: https://gitlab.freedesktop.org/drm/intel/-/issues/8307 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4ffa3a59f419..5c6980394dce 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4604,7 +4604,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862818, "Raptorlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), base-commit: be6247640eea9d9b0ff15607fab7a12f40974985 -- 2.40.0

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