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

[PATCH] ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs
by Stephan Gerhold 15 Apr '20

15 Apr '20

For some reason, the MI2S DAIs do not have channels_min/max defined. This means that snd_soc_dai_stream_valid() returns false, i.e. the DAIs have neither valid playback nor capture stream. It's quite surprising that this ever worked correctly, but in 5.7-rc1 this is now failing badly: :) Commit 0e9cf4c452ad ("ASoC: pcm: check if cpu-dai supports a given stream") introduced a check for snd_soc_dai_stream_valid() before calling hw_params(), which means that the q6i2s_hw_params() function was never called, eventually resulting in: qcom-q6afe aprsvc:q6afe:4:4: no line is assigned ... even though "qcom,sd-lines" is set in the device tree. Commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") now even avoids creating PCM devices if the stream is not supported, which means that it is failing even earlier with e.g.: Primary MI2S: ASoC: no backend playback stream Avoid all that trouble by adding channels_min/max for the MI2S DAIs. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..2a5302f1db98 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -902,6 +902,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -917,6 +919,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -931,6 +935,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -946,6 +952,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -960,6 +968,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -975,6 +985,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -989,6 +1001,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -1004,6 +1018,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, -- 2.26.1

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[PATCH v4 0/9] x86: Easy way of detecting MS Surface 3
by Andy Shevchenko 15 Apr '20

15 Apr '20

While working on RTC regression, I noticed that we are using the same DMI check over and over in the drivers for MS Surface 3 platform. This series dedicated for making it easier in the same way how it's done for Apple machines. Changelog v4: - rebase on top of v5.7-rc1 Changelog v3: - fixed typo in patch 5 (Jonathan) - returned back to if {} else {} condition in ASoC driver (Mark) - added Mark's Ack tag Changelog v2: - removed RTC patches for now (the fix will be independent to this series) - added couple more clean ups to arch/x86/kernel/quirks.c - redone DMI quirk to use driver_data instead of callback - simplified check in soc-acpi-intel-cht-match.c to be oneliner - added a new patch to cover rt5645 codec driver Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Andy Shevchenko (9): x86/platform: Rename x86/apple.h -> x86/machine.h x86/quirks: Add missed include to satisfy static checker x86/quirks: Introduce hpet_dev_print_force_hpet_address() helper x86/quirks: Join string literals back x86/quirks: Convert DMI matching to use a table x86/quirks: Add a DMI quirk for Microsoft Surface 3 platform/x86: surface3_wmi: Switch DMI table match to a test of variable ASoC: rt5645: Switch DMI table match to a test of variable ASoC: Intel: Switch DMI table match to a test of variable arch/x86/kernel/quirks.c | 91 +++++++++++++------ drivers/platform/x86/surface3-wmi.c | 16 +--- include/linux/platform_data/x86/apple.h | 14 +-- include/linux/platform_data/x86/machine.h | 20 ++++ sound/soc/codecs/rt5645.c | 14 ++- .../intel/common/soc-acpi-intel-cht-match.c | 28 +----- 6 files changed, 93 insertions(+), 90 deletions(-) create mode 100644 include/linux/platform_data/x86/machine.h -- 2.25.1

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[PATCH] ALSA: hda/hdmi: Add module option to disable audio component binding
by Takashi Iwai 15 Apr '20

15 Apr '20

As the recent regression showed, we want sometimes to turn off the audio component binding just for debugging. This patch adds the module option to control it easily without compilation. Fixes: ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bb287a916dae..4eff16053bd5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,10 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +static bool enable_acomp = true; +module_param(enable_acomp, bool, 0444); +MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)"); + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -2505,6 +2509,11 @@ static void generic_acomp_init(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; + if (!enable_acomp) { + codec_info(codec, "audio component disabled by module option\n"); + return; + } + spec->port2pin = port2pin; setup_drm_audio_ops(codec, ops); if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, -- 2.25.0

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[PATCH] bytcht-es8316: fix Speaker or Headphone no sound if enable Mic or Headset
by youling257 15 Apr '20

15 Apr '20

alsaucm -c bytcht-es8316 set _verb HiFi set _enadev Mic, Speaker will no sound. alsaucm -c bytcht-es8316 set _verb HiFi set _enadev Headset, Headphone will no sound. Signed-off-by: youling257 <youling257(a)gmail.com> --- ucm2/codecs/es8316/EnableSeq.conf | 2 -- 1 file changed, 2 deletions(-) diff --git a/ucm2/codecs/es8316/EnableSeq.conf b/ucm2/codecs/es8316/EnableSeq.conf index 551b79b..53bb115 100644 --- a/ucm2/codecs/es8316/EnableSeq.conf +++ b/ucm2/codecs/es8316/EnableSeq.conf @@ -1,6 +1,4 @@ # Disable all inputs / outputs -cset "name='Speaker Switch' off" -cset "name='Headphone Switch' off" cset "name='Headset Mic Switch' off" cset "name='Internal Mic Switch' off" -- 2.25.1

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Debugging dmix?
by Mark Hills 15 Apr '20

15 Apr '20

Recommendations for debugging dmix? For a long time I have a problem where audio output regresses to glitches, only solved by closing all applications using audio and reopening. Especially common when Chromium is used: 3-4 times a day, frustrating. I have a very explicit dmix configuration (see below); no PulseAudio. Reproducable on latest alsa-lib (fb48ad9e; later than v1.2.2) I can rule out _some_ aspects of hardware/drivers (Layla3G); eg. when the fault occurs, other soundcard channels are fine. Any way to understand what dmix is doing? Clients? Control process? Here's a recording taken from the analogue output (L+R chanel get zero'd): http://www.pogo.org.uk/~mark/tmp/dmix-glitch.png http://www.pogo.org.uk/~mark/tmp/dmix-glitch.flac Some process overwriting the ring buffer? Not high enough thread priority? Seems like it might be more prevalant with more applications using sound. Maybe why Chromium makes it worse; several threads/sandboxes, perhaps. dmix has all the trappings of something fiendishly clever and concise, but quite opaque. I'd need time to study the code (it has few comments, a lot of assumed knowledge) to understand its IPC. Can anyone give a head start? Specifically, how does the final mix get written to the output? eg. Thread priorities, if that process goes away etc. Thanks --- pcm.layla_multi { type route slave.pcm { type dmix ipc_key 833282 ipc_key_add_uid true slave { pcm "hw:Layla3G,0,0" period_size 256 buffer_size 32768 channels 4 rate 48000 format S24_3LE } } ttable { # Headphones 0.0 1 1.1 1 # Speakers 0.2 1 1.3 1 } } pcm.!default { type plug slave.pcm layla_multi } -- Mark

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[PATCH v6 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 15 Apr '20

15 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 21 +- sound/soc/fsl/fsl_asrc.c | 303 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 105 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2114 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3206 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

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[PATCH] ASoC: Intel: skl_hda_generic: remove rtd->codec_dai
by Pierre-Louis Bossart 15 Apr '20

15 Apr '20

Use macro and solve compilation issues Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 64197b010e7d..35f222ed9912 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -175,7 +175,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct hdac_hda_priv *hda_pvt; if (!codec_dai) -- 2.20.1

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[PATCH] ASoC: fsl_micfil: Omit superfluous error message in fsl_micfil_probe()
by Tang Bin 15 Apr '20

15 Apr '20

In the function fsl_micfil_probe(), when get irq failed, the function platform_get_irq() logs an error message, so remove redundant message here. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> Signed-off-by: Shengju Zhang <zhangshengju(a)cmss.chinamobile.com> --- sound/soc/fsl/fsl_micfil.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index f7f2d29f1..e73bd6570 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -702,10 +702,8 @@ static int fsl_micfil_probe(struct platform_device *pdev) for (i = 0; i < MICFIL_IRQ_LINES; i++) { micfil->irq[i] = platform_get_irq(pdev, i); dev_err(&pdev->dev, "GET IRQ: %d\n", micfil->irq[i]); - if (micfil->irq[i] < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (micfil->irq[i] < 0) return micfil->irq[i]; - } } if (of_property_read_bool(np, "fsl,shared-interrupt")) -- 2.20.1.windows.1

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[PATCH] ASoC: sgtl5000: Fix VAG power-on handling
by Sebastian Reichel 15 Apr '20

15 Apr '20

As mentioned slightly out of patch context in the code, there is no reset routine for the chip. On boards where the chip is supplied by a fixed regulator, it might not even be resetted during (e.g. watchdog) reboot and can be in any state. If the device is probed with VAG enabled, the driver's probe routine will generate a loud pop sound when ANA_POWER is being programmed. Avoid this by properly disabling just the VAG bit and waiting the required power down time. Signed-off-by: Sebastian Reichel <sebastian.reichel(a)collabora.com> --- sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 1 + 2 files changed, 35 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d5130193b4a2..e8a8bf7b4ffe 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "Error %d initializing CHIP_CLK_CTRL\n", ret); + /* Mute everything to avoid pop from the following power-up */ + ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_CHIP_ANA_CTRL_DEFAULT); + if (ret) { + dev_err(&client->dev, + "Error %d muting outputs via CHIP_ANA_CTRL\n", ret); + goto disable_clk; + } + + /* + * If VAG is powered-on (e.g. from previous boot), it would be disabled + * by the write to ANA_POWER in later steps of the probe code. This + * may create a loud pop even with all outputs muted. The proper way + * to circumvent this is disabling the bit first and waiting the proper + * cool-down time. + */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value); + if (ret) { + dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret); + goto disable_clk; + } + if (value & SGTL5000_VAG_POWERUP) { + ret = regmap_update_bits(sgtl5000->regmap, + SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, + 0); + if (ret) { + dev_err(&client->dev, "Error %d disabling VAG\n", ret); + goto disable_clk; + } + + msleep(SGTL5000_VAG_POWERDOWN_DELAY); + } + /* Follow section 2.2.1.1 of AN3663 */ ana_pwr = SGTL5000_ANA_POWER_DEFAULT; if (sgtl5000->num_supplies <= VDDD) { diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index a4bf4bca95bf..56ec5863f250 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -233,6 +233,7 @@ /* * SGTL5000_CHIP_ANA_CTRL */ +#define SGTL5000_CHIP_ANA_CTRL_DEFAULT 0x0133 #define SGTL5000_LINE_OUT_MUTE 0x0100 #define SGTL5000_HP_SEL_MASK 0x0040 #define SGTL5000_HP_SEL_SHIFT 6 -- 2.25.1

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[PATCH 1/2] dt-bindings: sound: lpass-cpu: Document DAI subnodes
by Stephan Gerhold 15 Apr '20

15 Apr '20

The lpass-cpu driver now allows configuring the MI2S SD lines by defining subnodes for one of the DAIs. Document this in the device tree bindings. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- .../bindings/sound/qcom,lpass-cpu.txt | 23 +++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index 21c648328be9..df53a10502f7 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -30,6 +30,8 @@ Required properties: - reg : Must contain an address for each entry in reg-names. - reg-names : A list which must include the following entries: * "lpass-lpaif" +- #address-cells : Must be 1 +- #size-cells : Must be 0 @@ -37,6 +39,18 @@ Optional properties: - qcom,adsp : Phandle for the audio DSP node +By default, the driver uses up to 4 MI2S SD lines, for a total of 8 channels. +The SD lines to use can be configured by adding subnodes for each of the DAIs. + +Required properties for each DAI (represented by a subnode): +- reg : Must be one of the DAI IDs + (usually part of dt-bindings header) +- qcom,playback-sd-lines: List of serial data lines (0-3) to use for playback +- qcom,capture-sd-lines : List of serial data lines (0-3) to use for capture + +Note that adding a subnode changes the default to "no lines configured", +so both playback and capture lines should be configured when a subnode is added. + Example: lpass@28100000 { @@ -51,4 +65,13 @@ lpass@28100000 { reg = <0x28100000 0x10000>; reg-names = "lpass-lpaif"; qcom,adsp = <&adsp>; + + #address-cells = <1>; + #size-cells = <0>; + + /* Optional to set different MI2S SD lines */ + mi2s-quaternary@3 { + reg = <MI2S_QUATERNARY>; + qcom,playback-sd-lines = <0 1>; + }; }; -- 2.26.0

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