June 2020 - Alsa-devel - mailman.alsa-project.org

sof-hda-dsp: make Headphone Playback Switch conditional
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-ucm-conf pull request #34 was opened from kv2019i: The Headphone Playback Switch control is not present in all HDA codecs. Allow the Headphones definition to work also on such systems. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/34 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/34.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Clock sync problem?
by Laurence Tratt 11 Jun '20

11 Jun '20

Hello, I've found an interesting issue with an SSL2+ audio interface -- it plays and records fine (and, for the money, the sound quality is excellent!), but recording slowly but surely drifts over time. For example, if I record a click track in Audacity (or Ardour), play it through the interface and record it, and then line up the start of the recorded click track with the original, the two tracks will be around 5ms out of alignment after 30s. Interestingly, this problem does not happen with OpenBSD, where the two tracks line up precisely. My guess is that the problem is to do with implicit feedback in endpoints. I wondered if the problem was the same as with the MOTU M4, but enabling that quirk for the SSL2+ introduced some crackle, but didn't change the drift. I'm attaching a) the alsa-info file b) output from OpenBSD's kernel with debugging in the USB audio driver turned on in case that gives some clues as to what the two OS's and drivers might be doing differently. If anyone has any suggestions for how this problem might be addressed, I would be very grateful! Laurie -- Personal http://tratt.net/laurie/ Software Development Team http://soft-dev.org/ https://github.com/ltratt http://twitter.com/laurencetratt

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alsa-gobject: obsolete abstract class for `snd_timer_read` and `snd_timer_tread` structure
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-gobject pull request #31 was opened from takaswie: In UAPI of Linux sound subsystem, two types of event is read from ALSA Timer character device; snd_timer_read and snd_timer_tread. Before attached to any timer device, an user instance can select which type of event is expected. ALSATimer.UserInstance uses one GObject signal to handle these events. These events are defined as derived object from abstract class, ALSATimer.EventData. However, the design pattern is not convenient for some language bindings. For example, in Rust language, class-based inheritance is not supported. Gtk-rs project produces mechanism to upcast/downcast between super/sub classes and handles GObject class system. In my opinion, the code to upcast/downcast is a bit complicated to handle derivatives. This patchset obsoletes usage of abstract class for ALSATimer.EventData. Instead, ALSATimer.Event is defined as concrete class and the instance of class is passed to the GObject signal. Applications use APIs to pick up expected type of data from the instance. Request URL : https://github.com/alsa-project/alsa-gobject/pull/31 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[RFC PATCH] ALSA: compress: Fix gapless playback state machine
by Srinivas Kandagatla 11 Jun '20

11 Jun '20

For gapless playback call to snd_compr_drain_notify() after partial drain should put the state to SNDRV_PCM_STATE_RUNNING rather than SNDRV_PCM_STATE_SETUP as the driver is ready to process the buffers for new track. With existing code, if we are playing 3 tracks in gapless, after partial drain finished on previous track 1 the state is set to SNDRV_PCM_STATE_SETUP which is then moved to SNDRV_PCM_STATE_PREPARED after data write. With this state calls to snd_compr_next_track() and few other calls will fail as they expect the state to be in SNDRV_PCM_STATE_RUNNING. Here is the sequence of events and state transitions: 1. set_params (Track 1), state = SNDRV_PCM_STATE_SETUP 2. set_metadata (Track 1), no state change, state = SNDRV_PCM_STATE_SETUP 3. fill and trigger start (Track 1), state = SNDRV_PCM_STATE_RUNNING 4. set_next_track (Track 2), state = SNDRV_PCM_STATE_RUNNING 5. partial_drain (Track 1), state = SNDRV_PCM_STATE_SETUP 6 snd_compr_drain_notify (Track 1), state = SNDRV_PCM_STATE_SETUP 7. fill data (Track 2), state = SNDRV_PCM_STATE_PREPARED 8. set_metadata (Track 3), no state change, state = SNDRV_PCM_STATE_PREPARED 9. set_next_track (Track 3), !! FAILURE as state != SNDRV_PCM_STATE_RUNNING Fixes: f44f2a5417b2 ("ALSA: compress: fix drain calls blocking other compress functions (v6)") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- I wonder who did gapless work on upstream so far? include/sound/compress_driver.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 6ce8effa0b12..eabac33864c2 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -182,7 +182,7 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) if (snd_BUG_ON(!stream)) return; - stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; wake_up(&stream->runtime->sleep); } -- 2.21.0

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[PATCH 2/2] ASoC: max98390: Update regmap readable reg and volatile
by Steve Lee 11 Jun '20

11 Jun '20

Update max98390_readable_register and max98390_volatile_reg Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 57d15dbfcda3..b345e626956d 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -700,8 +700,8 @@ static bool max98390_readable_register(struct device *dev, unsigned int reg) case MAX98390_IRQ_CTRL ... MAX98390_WDOG_CTRL: case MAX98390_MEAS_ADC_THERM_WARN_THRESH ... MAX98390_BROWNOUT_INFINITE_HOLD: - case MAX98390_BROWNOUT_LVL_HOLD ... THERMAL_COILTEMP_RD_BACK_BYTE0: - case DSMIG_DEBUZZER_THRESHOLD ... MAX98390_R24FF_REV_ID: + case MAX98390_BROWNOUT_LVL_HOLD ... DSMIG_DEBUZZER_THRESHOLD: + case DSM_VOL_ENA ... MAX98390_R24FF_REV_ID: return true; default: return false; @@ -717,7 +717,7 @@ static bool max98390_volatile_reg(struct device *dev, unsigned int reg) case MAX98390_BROWNOUT_LOWEST_STATUS: case MAX98390_ENV_TRACK_BOOST_VOUT_READ: case DSM_STBASS_HPF_B0_BYTE0 ... DSM_DEBUZZER_ATTACK_TIME_BYTE2: - case THERMAL_RDC_RD_BACK_BYTE1 ... THERMAL_COILTEMP_RD_BACK_BYTE0: + case THERMAL_RDC_RD_BACK_BYTE1 ... DSMIG_DEBUZZER_THRESHOLD: case DSM_THERMAL_GAIN ... DSM_WBDRC_GAIN: return true; default: -- 2.17.1

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[PATCH 1/2] ASoC: max98390: Add Amp init common setting func.
by Steve Lee 11 Jun '20

11 Jun '20

Add amp common init function to gather common init setting and finaize. - add max98390_init_regs func - move amp setting to max98390_init_regs func. - removed unneceary setting and finalize common register values. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 0d63ebfbff2f..57d15dbfcda3 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -842,6 +842,20 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) return 0; } +static void max98390_init_regs(struct snd_soc_component *component) +{ + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + regmap_write(max98390->regmap, MAX98390_CLK_MON, 0x6f); + regmap_write(max98390->regmap, MAX98390_DAT_MON, 0x00); + regmap_write(max98390->regmap, MAX98390_PWR_GATE_CTL, 0x00); + regmap_write(max98390->regmap, MAX98390_PCM_RX_EN_A, 0x03); + regmap_write(max98390->regmap, MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0e); + regmap_write(max98390->regmap, MAX98390_BOOST_BYPASS1, 0x46); + regmap_write(max98390->regmap, MAX98390_FET_SCALING3, 0x03); +} + static int max98390_probe(struct snd_soc_component *component) { struct max98390_priv *max98390 = @@ -853,18 +867,10 @@ static int max98390_probe(struct snd_soc_component *component) /* Update dsm bin param */ max98390_dsm_init(component); - /* Amp Setting */ - regmap_write(max98390->regmap, MAX98390_CLK_MON, 0x6f); - regmap_write(max98390->regmap, MAX98390_PCM_RX_EN_A, 0x03); - regmap_write(max98390->regmap, MAX98390_PWR_GATE_CTL, 0x2d); - regmap_write(max98390->regmap, MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0e); - regmap_write(max98390->regmap, MAX98390_BOOST_BYPASS1, 0x46); - regmap_write(max98390->regmap, MAX98390_FET_SCALING3, 0x03); + /* Amp init setting */ + max98390_init_regs(component); /* Dsm Setting */ - regmap_write(max98390->regmap, DSM_VOL_CTRL, 0x94); - regmap_write(max98390->regmap, DSMIG_EN, 0x19); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); if (max98390->ref_rdc_value) { regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE0, max98390->ref_rdc_value & 0x000000ff); -- 2.17.1

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[PATCH v2 0/3] ASoC: ti: Add support for audio on J721e EVM
by Peter Ujfalusi 11 Jun '20

11 Jun '20

Hi, Changes since v1: - Fixed DT binding documentation errors - Rebased on ASoC head and updated the driver to compile and work This series adds support for the analog audio setup on the j721e EVM. The audio setup of the EVM is: Common Processor Board (CPB): McASP10 <-> pcm3168a Infotainment Expansion Board (IVI): McASP0 <-> 2x pcm3168a Both CPB and IVI wired in parallel serializer setup. The first patch adds the stream_name for McASP driver as it is needed in multicodec (and would be needed in DPCM) setup for proper DAPM handling. The second patch adds two DT schema, one for the cpb and one for the cpb+ivi card. Regards, Peter --- Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture bindings: sound: Add documentation for TI j721e EVM (CPB and IVI) ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI) .../bindings/sound/ti,j721e-cpb-audio.yaml | 93 ++ .../sound/ti,j721e-cpb-ivi-audio.yaml | 142 +++ sound/soc/ti/Kconfig | 8 + sound/soc/ti/Makefile | 2 + sound/soc/ti/davinci-mcasp.c | 3 + sound/soc/ti/j721e-evm.c | 867 ++++++++++++++++++ 6 files changed, 1115 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml create mode 100644 sound/soc/ti/j721e-evm.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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alsa-gobject: obsolete abstract class for snd_seq_queue_timer structure
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-gobject pull request #30 was opened from takaswie: In UAPI of Linux sound subsystem, `snd_seq_queue_timer` structure includes union member. Current implementation of ALSASeq uses abstract class (ALSASeqQueueTimer) and concrete class (ALSASeqQueueTimerAlsa) to represent variation corresponding to each of union member. However, the design pattern is not convenient for some language bindings. For example, in Rust language, class-based inheritance is not supported. Gtk-rs project produces mechanism to upcast/downcast between super/sub classes and handles GObject class system. In my opinion, the code to upcast/downcast is a bit complicated to handle derivatives. This patchset obsoletes usage of abstract class for ALSASeq. The target is ALSASeqQueueTimer. ALSASeqQueueTimerDataAlsa is newly defined to retrieve data for the case of ALSATimer and the inherited ALSASeqQueueTimerAlsa is deleted. Request URL : https://github.com/alsa-project/alsa-gobject/pull/30 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/30.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-gobject: seq: misc fixes (take 2)
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-gobject pull request #29 was opened from takaswie: This patchset includes misc fixes for ALSASeq. Especially the execution of g-ir-scanner to generate ALSASeq-0.0 becomes to expect ALSATimer-0.0 is available. Although it is expected, it results in Python 3 gir loader (PyGObject) loading ALSATimer-0.0 before loading ALSASeq-0.0. Any test for ALSASeq-0.0 is going to fail because ALSATimer-0.0 is not available before installing. To solve the issue, the patchset includes the patch to change test procedure to arrange environment variables. Request URL : https://github.com/alsa-project/alsa-gobject/pull/29 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/29.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 1/2] dt-bindings: ASoc: Fix tdm-slot documentation spelling error
by Dan Murphy 10 Jun '20

10 Jun '20

Fix the spelling of 'specified'. Also fix grammarical issue with the use of 'a' over 'an' Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tdm-slot.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt index 34cf70e2cbc4..4bb513ae62fc 100644 --- a/Documentation/devicetree/bindings/sound/tdm-slot.txt +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -14,8 +14,8 @@ For instance: dai-tdm-slot-tx-mask = <0 1>; dai-tdm-slot-rx-mask = <1 0>; -And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() -to specify a explicit mapping of the channels and the slots. If it's absent +And for each specified driver, there could be one .of_xlate_tdm_slot_mask() +to specify an explicit mapping of the channels and the slots. If it's absent the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the tx and rx masks. -- 2.26.2

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