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[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

3 2

[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

1 0

[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

1 0

[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

4 22

[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

2 1

[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

2 1

[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

3 2

[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

2 3

[PATCH] ALSA: usb-audio: RME Babyface Pro mixer patch
by Thomas Ebeling 15 Apr '20

15 Apr '20

Added mixer quirks to allow controlling the internal DSP of the RME Babyface Pro and its successor Babyface Pro FS. Signed-off-by: Thomas Ebeling <penguins(a)bollie.de> --- sound/usb/mixer_quirks.c | 418 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 418 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 02b036b2aefb..f0302ad7f5a0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2181,6 +2181,421 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * RME Babyface Pro (FS) + * + * These devices exposes a couple of DSP functions via request to EP0. + * Switches are available via control registers, while routing is controlled + * by controlling the volume on each possible crossing point. + * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with + * 0dB being at dec. 32768. + */ +enum { + SND_BBFPRO_CTL_REG1 = 0, + SND_BBFPRO_CTL_REG2 +}; + +#define SND_BBFPRO_CTL_REG_MASK 1 +#define SND_BBFPRO_CTL_IDX_MASK 0xff +#define SND_BBFPRO_CTL_IDX_SHIFT 1 +#define SND_BBFPRO_CTL_VAL_MASK 1 +#define SND_BBFPRO_CTL_VAL_SHIFT 9 +#define SND_BBFPRO_CTL_REG1_CLK_MASTER 0 +#define SND_BBFPRO_CTL_REG1_CLK_OPTICAL 1 +#define SND_BBFPRO_CTL_REG1_SPDIF_PRO 7 +#define SND_BBFPRO_CTL_REG1_SPDIF_EMPH 8 +#define SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL 10 +#define SND_BBFPRO_CTL_REG2_48V_AN1 0 +#define SND_BBFPRO_CTL_REG2_48V_AN2 1 +#define SND_BBFPRO_CTL_REG2_SENS_IN3 2 +#define SND_BBFPRO_CTL_REG2_SENS_IN4 3 +#define SND_BBFPRO_CTL_REG2_PAD_AN1 4 +#define SND_BBFPRO_CTL_REG2_PAD_AN2 5 + +#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff +#define SND_BBFPRO_MIXER_VAL_SHIFT 9 +#define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf +#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB + +#define SND_BBFPRO_USBREQ_CTL_REG1 0x10 +#define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_MIXER 0x12 + +static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, + u8 index, u8 value) +{ + int err; + u16 usb_req, usb_idx, usb_val; + struct snd_usb_audio *chip = mixer->chip; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + if (reg == SND_BBFPRO_CTL_REG1) { + usb_req = SND_BBFPRO_USBREQ_CTL_REG1; + if (index == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + usb_idx = 3; + usb_val = value ? 3 : 0; + } else { + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + } else { + usb_req = SND_BBFPRO_USBREQ_CTL_REG2; + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), usb_req, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 reg, idx, val; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + val = kcontrol->private_value >> SND_BBFPRO_CTL_VAL_SHIFT; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + ucontrol->value.enumerated.item[0] = val; + } else { + ucontrol->value.integer.value[0] = val; + } + return 0; +} + +static int snd_bbfpro_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + u8 reg, idx; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + + if (reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + static const char * const texts[2] = { + "AutoSync", + "Internal" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } else if (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4)) { + static const char * const texts[2] = { + "-10dBV", + "+4dBu" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + return 0; +} + +static int snd_bbfpro_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u8 reg, idx; + int old_value, pv, val; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + old_value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + val = ucontrol->value.enumerated.item[0]; + } else { + val = ucontrol->value.integer.value[0]; + } + + if (val > 1) + return -EINVAL; + + if (val == old_value) + return 0; + + kcontrol->private_value = reg + | ((idx & SND_BBFPRO_CTL_IDX_MASK) << SND_BBFPRO_CTL_IDX_SHIFT) + | ((val & SND_BBFPRO_CTL_VAL_MASK) << SND_BBFPRO_CTL_VAL_SHIFT); + + err = snd_bbfpro_ctl_update(mixer, reg, idx, val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) +{ + u8 reg, idx; + int value, pv; + + pv = list->kctl->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); +} + +static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, + u32 value) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + u16 idx; + u16 usb_idx, usb_val; + u32 v; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + idx = index & SND_BBFPRO_MIXER_IDX_MASK; + // 18 bit linear volume, split so 2 bits end up in index. + v = value & SND_BBFPRO_MIXER_VAL_MASK; + usb_idx = idx | (v & 0x3) << 14; + usb_val = (v >> 2) & 0xffff; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_MIXER, + USB_DIR_OUT | USB_TYPE_VENDOR | + USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + return 0; +} + +static int snd_bbfpro_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_MIXER_VAL_MIN; + uinfo->value.integer.max = SND_BBFPRO_MIXER_VAL_MAX; + return 0; +} + +static int snd_bbfpro_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u16 idx; + u32 new_val, old_value, uvalue; + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + uvalue = ucontrol->value.integer.value[0]; + idx = kcontrol->private_value & SND_BBFPRO_MIXER_IDX_MASK; + old_value = kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + + if (uvalue > SND_BBFPRO_MIXER_VAL_MAX) + return -EINVAL; + + if (uvalue == old_value) + return 0; + + new_val = uvalue & SND_BBFPRO_MIXER_VAL_MASK; + + kcontrol->private_value = idx + | (new_val << SND_BBFPRO_MIXER_VAL_SHIFT); + + err = snd_bbfpro_vol_update(mixer, idx, new_val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_vol_resume(struct usb_mixer_elem_list *list) +{ + int pv = list->kctl->private_value; + u16 idx = pv & SND_BBFPRO_MIXER_IDX_MASK; + u32 val = (pv >> SND_BBFPRO_MIXER_VAL_SHIFT) + & SND_BBFPRO_MIXER_VAL_MASK; + return snd_bbfpro_vol_update(list->mixer, idx, val); +} + +// Predfine elements +static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_ctl_info, + .get = snd_bbfpro_ctl_get, + .put = snd_bbfpro_ctl_put +}; + +static const struct snd_kcontrol_new snd_bbfpro_vol_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_vol_info, + .get = snd_bbfpro_vol_get, + .put = snd_bbfpro_vol_put +}; + +static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, + u8 index, char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_ctl_control; + + knew.name = name; + knew.private_value = (reg & SND_BBFPRO_CTL_REG_MASK) + | ((index & SND_BBFPRO_CTL_IDX_MASK) + << SND_BBFPRO_CTL_IDX_SHIFT); + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_ctl_resume, + &knew, NULL); +} + +static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_vol_control; + + knew.name = name; + knew.private_value = index & SND_BBFPRO_MIXER_IDX_MASK; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_vol_resume, + &knew, NULL); +} + +static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i, o; + char name[48]; + + static const char * const input[] = { + "AN1", "AN2", "IN3", "IN4", "AS1", "AS2", "ADAT3", + "ADAT4", "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + static const char * const output[] = { + "AN1", "AN2", "PH3", "PH4", "AS1", "AS2", "ADAT3", "ADAT4", + "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + for (o = 0 ; o < 12 ; ++o) { + for (i = 0 ; i < 12 ; ++i) { + // Line routing + snprintf(name, sizeof(name), + "%s-%s-%s Playback Volume", + (i < 2 ? "Mic" : "Line"), + input[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + i), name); + if (err < 0) + return err; + + // PCM routing... yes, it is output remapping + snprintf(name, sizeof(name), + "PCM-%s-%s Playback Volume", + output[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + 12 + i), + name); + if (err < 0) + return err; + } + } + + // Control Reg 1 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_CLK_OPTICAL, + "Sample Clock Source"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_PRO, + "IEC958 Pro Mask"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_EMPH, + "IEC958 Emphasis"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL, + "IEC958 Switch"); + if (err < 0) + return err; + + // Control Reg 2 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN1, + "Mic-AN1 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN2, + "Mic-AN2 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN3, + "Line-IN3 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN4, + "Line-IN4 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN1, + "Mic-AN1 PAD"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN2, + "Mic-AN2 PAD"); + if (err < 0) + return err; + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2282,6 +2697,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ err = snd_sc1810_init_mixer(mixer); break; + case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ + err = snd_bbfpro_controls_create(mixer); + break; } return err; -- 2.26.0

2 1

monotonic raw setup seems buggy
by sylvain.bertrand＠gmail.com 15 Apr '20

15 Apr '20

Hi, I am writting an alsa application and while working on pcm status timings, I noticed that setting monotonic_raw timings in sw_params while the hardware device plugin is already running with monotonic timings did not produce an error. Follow the pcm dump which shows monotonic_raw being set on all plugin instances but the already running hw plugin instance. Is this a bug, or am I doing something the wrong way? Plug PCM: Linear Integer <-> Linear Float conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_NONINTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC_RAW period_step : 1 avail_min : 1024 period_event : 1 start_threshold : 1 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC_RAW period_step : 1 avail_min : 1024 period_event : 1 start_threshold : 1 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 1484839806 cheers, -- Sylvain

5 20

[PATCH] ALSA: hda: enable the runtime_pm for non-vgaswitcheroo hda controllers
by Hui Wang 14 Apr '20

14 Apr '20

Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbib() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled even it calls put_autosuspend(). Need to call runtime_allow() for those controllers in the hda driver. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8519051a426e..779705bef88b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2356,6 +2356,8 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + if (!use_vga_switcheroo(chip)) + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.17.1

4 11

[FRUSTRATED] Missing Sound Codecs for Lenovo ThinkCentre M710e
by Michael 14 Apr '20

14 Apr '20

Hi All, Technically this probably should go to a support channnel, and not a development list, but at my wits end. Kernel: 4.15.0-91-generic #92~16.04.1-Ubuntu SMP dmesg | grep snd [ 12.656843] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 12.807521] snd_hda_intel 0000:00:1f.3: CORB reset timeout#1, CORBRP = 0 [ 12.810429] snd_hda_intel 0000:00:1f.3: no codecs found! I have been playing with settings in the modprobe directory for Also with no effect.. current settings are.. options snd-hda-intel single_cmd=1 options snd-hda-intel probe_mask=1 options snd-hda-intel model=thinkpad options snd-hda-intel position_fix=3 Have tried model=generic, and various other suggestions from online, but at a loss. Don't see anything in: https://github.com/torvalds/linux/blob/master/Documentation/sound/hd-audio/… Your ALSA information is located at http://alsa-project.org/db/?f=7898b30139e9a91fecec22f3ee46049f33248b85 Can anyone point me to how to debug this further, or any way to force detection of a proper codec for this model? 00:1f.3 Audio device [0403]: Intel Corporation 200 Series PCH HD Audio [8086:a2f0] Subsystem: Lenovo Device [17aa:313c] Kernel driver in use: snd_hda_intel description: Motherboard product: 313C vendor: LENOVO physical id: 0 version: SDK0J40697 WIN 3305181519567 slot: Default string Booting in legacy OS mode, not EUFI.. -- "Catch the Magic of Linux..." ------------------------------------------------------------------------ Michael Peddemors, President/CEO LinuxMagic Inc. Visit us at http://www.linuxmagic.com @linuxmagic A Wizard IT Company - For More Info http://www.wizard.ca "MagicSpam" is a Registered TradeMark of Wizard Tower TechnoServices Ltd. ------------------------------------------------------------------------ 604-682-0300 Beautiful British Columbia, Canada This email and any electronic data contained are confidential and intended solely for the use of the individual or entity to which they are addressed. Please note that any views or opinions presented in this email are solely those of the author and are not intended to represent those of the company.

3 5

[PATCH 00/13] ASoC: Intel: machine drivers update for 5.8
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

This patchset a) adds support for the Intel ElkhartLake platforms b)aligns the HDaudio mic detection with the snd-hda-intel driver c) correct DMIC missing configurations d) fixes initialization/compilation problems for SoundWire platforms d) completes the removal of codec_dais missing in Morimoto-san's series. Bard Liao (1): ASoC: intel: sof_sdw: init all aggregated codecs Hui Wang (1): ASoC: intel/skl/hda - set autosuspend timeout for hda codecs Keyon Jie (1): ASoC: Intel: sof-da7219-max98373: add DMIC widget and route Libin Yang (3): ALSA: hda: Add ElkhartLake HDMI codec vid ASoC: SOF: Intel: add PCI ID for ElkhartLake ASoC: Intel: boards: support Elkhart Lake with rt5660 Pierre-Louis Bossart (6): ASoC: Intel: sof_sdw_hdmi: fix compilation issue in fallback mode ASoC: Intel: sof_sdw_hdmi: remove codec_dai use ASoC: Intel: sof_sdw_rt1308: remove codec dai use ASoC: Intel: sof_sdw_rt5682: remove codec_dai use ASoC: Intel: sof_sdw_rt700: remove codec_dai use ASoC: Intel: sof_sdw_rt711: remove codec_dai use Yong Zhi (1): ASoC: Intel: sof_da7219_max98373: Add BE dailink for dmic16k sound/pci/hda/patch_hdmi.c | 1 + sound/soc/intel/boards/Kconfig | 15 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/ehl_rt5660.c | 323 ++++++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 29 +- sound/soc/intel/boards/sof_da7219_max98373.c | 21 ++ sound/soc/intel/boards/sof_sdw.c | 41 ++- sound/soc/intel/boards/sof_sdw_hdmi.c | 4 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 3 +- sound/soc/intel/boards/sof_sdw_rt700.c | 3 +- sound/soc/intel/boards/sof_sdw_rt711.c | 3 +- .../intel/common/soc-acpi-intel-ehl-match.c | 7 + sound/soc/sof/sof-pci-dev.c | 2 + 14 files changed, 434 insertions(+), 22 deletions(-) create mode 100644 sound/soc/intel/boards/ehl_rt5660.c base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

4 30

[PATCH v2] ALSA: hda: call runtime_allow() for all hda controllers
by Hui Wang 14 Apr '20

14 Apr '20

Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbid() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled because the runtime_allow() is not called. Since it is no harm calling runtime_allow() twice, here let hda driver call runtime_allow() for all controllers. Then the runtime_pm is enabled on all controllers after the put_autosuspend() is called. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8519051a426e..a5fab12defde 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2356,6 +2356,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.17.1

2 1

[PATCH 0/2] ASoC: SOF: trivial code cleanups
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

Fix warnings reported by tools - no functionality change. Payal Kshirsagar (2): ASoC: SOF: remove unneeded variables ASoC: SOF: Intel: hda: remove unnecessary parentheses sound/soc/sof/intel/hda-codec.c | 5 +---- sound/soc/sof/intel/hda-dsp.c | 8 ++++---- sound/soc/sof/nocodec.c | 6 ++---- 3 files changed, 7 insertions(+), 12 deletions(-) base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

2 4

[PATCH 0/4] ASoC: SOF: adjust dmesg verbosity
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

Decrease the dmesg verbosity to remove unnecessary logs on SoundWire platforms, and conversely add more information to help the community and downstream distros with HDaudio/SOF support (DMIC detection and card instanciation are the most prevalent issues on GitHub). Pierre-Louis Bossart (3): ASoC: codecs: rt1308-sdw: reduce verbosity ASoC: SOF: Intel: hda: reduce verbosity on SoundWire detection ASoC: SOF: Intel: hda: log number of microphones detected in NHLT tables Ranjani Sridharan (1): ASoC: soc-core: Add dynamic debug logs in soc_dai_link_sanity_check() sound/soc/codecs/rt1308-sdw.c | 4 ++-- sound/soc/soc-core.c | 18 +++++++++++++++--- sound/soc/sof/intel/hda.c | 10 ++++++---- 3 files changed, 23 insertions(+), 9 deletions(-) base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

2 8

Applied "ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c2233a266178f8937cc26a84cd7672334b5424b7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:47:37 +0900 Subject: [PATCH] ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87imimboli.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 12 ++++----- sound/soc/soc-compress.c | 36 +++++++++++++-------------- sound/soc/soc-core.c | 16 ++++++------ sound/soc/soc-dapm.c | 4 +-- sound/soc/soc-generic-dmaengine-pcm.c | 6 ++--- sound/soc/soc-pcm.c | 30 +++++++++++----------- 6 files changed, 52 insertions(+), 52 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 13458e4fbb13..4dff6745b067 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1179,16 +1179,16 @@ struct snd_soc_pcm_runtime { (i)++) #define for_each_rtd_cpu_dais(rtd, i, dai) \ for ((i) = 0; \ - ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]); \ + ((i) < rtd->num_cpus) && ((dai) = asoc_rtd_to_cpu(rtd, i)); \ (i)++) #define for_each_rtd_cpu_dais_rollback(rtd, i, dai) \ - for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);) + for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_cpu(rtd, i));) #define for_each_rtd_codec_dais(rtd, i, dai) \ for ((i) = 0; \ - ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ + ((i) < rtd->num_codecs) && ((dai) = asoc_rtd_to_codec(rtd, i)); \ (i)++) #define for_each_rtd_codec_dais_rollback(rtd, i, dai) \ - for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) + for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_codec(rtd, i));) #define for_each_rtd_dais(rtd, i, dai) \ for ((i) = 0; \ ((i) < (rtd)->num_cpus + (rtd)->num_codecs) && \ @@ -1381,8 +1381,8 @@ struct snd_soc_dai *snd_soc_card_get_codec_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd; list_for_each_entry(rtd, &card->rtd_list, list) { - if (!strcmp(rtd->codec_dai->name, dai_name)) - return rtd->codec_dai; + if (!strcmp(asoc_rtd_to_codec(rtd, 0)->name, dai_name)) + return asoc_rtd_to_codec(rtd, 0); } return NULL; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 50062eb79adb..3f77c0208e09 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -73,7 +73,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component, *save = NULL; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret, i; for_each_rtd_components(rtd, i, component) { @@ -141,7 +141,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) struct snd_pcm_substream *fe_substream = fe->pcm->streams[cstream->direction].substream; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list; int stream; @@ -230,8 +230,8 @@ static int soc_compr_free(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int stream, i; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -274,7 +274,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) static int soc_compr_free_fe(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); struct snd_soc_dpcm *dpcm; int stream, ret; @@ -343,8 +343,8 @@ static int soc_compr_components_trigger(struct snd_compr_stream *cstream, static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -373,7 +373,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); int ret, stream; if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN || @@ -446,7 +446,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -500,7 +500,7 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *fe = cstream->private_data; struct snd_pcm_substream *fe_substream = fe->pcm->streams[cstream->direction].substream; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); int ret, stream; if (cstream->direction == SND_COMPRESS_PLAYBACK) @@ -558,7 +558,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -632,7 +632,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -664,7 +664,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; int i, ret = 0; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -711,7 +711,7 @@ static int soc_compr_set_metadata(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret; if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_metadata) { @@ -739,7 +739,7 @@ static int soc_compr_get_metadata(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret; if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_metadata) { @@ -801,8 +801,8 @@ static struct snd_compr_ops soc_compr_dyn_ops = { int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_component *component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_compr *compr; struct snd_pcm *be_pcm; char new_name[64]; @@ -891,7 +891,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { - component = rtd->codec_dai->component; + component = asoc_rtd_to_codec(rtd, 0)->component; dev_err(component->dev, "Compress ASoC: can't create compress for codec %s: %d\n", component->name, ret); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 843b8b1c89d4..4ad64b96d893 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -364,7 +364,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); */ void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int playback = SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -991,13 +991,13 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, rtd->num_cpus = dai_link->num_cpus; for_each_link_cpus(dai_link, i, cpu) { - rtd->cpu_dais[i] = snd_soc_find_dai(cpu); - if (!rtd->cpu_dais[i]) { + asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu); + if (!asoc_rtd_to_cpu(rtd, i)) { dev_info(card->dev, "ASoC: CPU DAI %s not registered\n", cpu->dai_name); goto _err_defer; } - snd_soc_rtd_add_component(rtd, rtd->cpu_dais[i]->component); + snd_soc_rtd_add_component(rtd, asoc_rtd_to_cpu(rtd, i)->component); } /* Single cpu links expect cpu and cpu_dai in runtime data */ @@ -1006,14 +1006,14 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, /* Find CODEC from registered CODECs */ rtd->num_codecs = dai_link->num_codecs; for_each_link_codecs(dai_link, i, codec) { - rtd->codec_dais[i] = snd_soc_find_dai(codec); - if (!rtd->codec_dais[i]) { + asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec); + if (!asoc_rtd_to_codec(rtd, i)) { dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n", codec->dai_name); goto _err_defer; } - snd_soc_rtd_add_component(rtd, rtd->codec_dais[i]->component); + snd_soc_rtd_add_component(rtd, asoc_rtd_to_codec(rtd, i)->component); } /* Single codec links expect codec and codec_dai in runtime data */ @@ -1062,7 +1062,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai_link *dai_link = rtd->dai_link; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_component *component; int ret, num, i; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 679ed60d850e..8f260994b149 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4370,11 +4370,11 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, if (rtd->num_cpus == 1) { for_each_rtd_codec_dais(rtd, i, codec_dai) dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[0]); + asoc_rtd_to_cpu(rtd, 0)); } else if (rtd->num_codecs == rtd->num_cpus) { for_each_rtd_codec_dais(rtd, i, codec_dai) dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[i]); + asoc_rtd_to_cpu(rtd, i)); } else { dev_err(card->dev, "N cpus to M codecs link is not supported yet\n"); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index facf1922a714..f728309a0833 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -68,7 +68,7 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, return -EINVAL; } - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) @@ -134,7 +134,7 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); memset(&hw, 0, sizeof(hw)); hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -203,7 +203,7 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return NULL; } - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 289aebc15529..454735f8fa92 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -136,7 +136,7 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, return -ENOMEM; for_each_pcm_streams(stream) - if (snd_soc_dai_stream_valid(fe->cpu_dai, stream)) + if (snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream)) offset += dpcm_show_state(fe, stream, buf + offset, out_count - offset); @@ -836,10 +836,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) soc_pcm_init_runtime_hw(substream); if (rtd->num_codecs == 1) - codec_dai_name = rtd->codec_dai->name; + codec_dai_name = asoc_rtd_to_codec(rtd, 0)->name; if (rtd->num_cpus == 1) - cpu_dai_name = rtd->cpu_dai->name; + cpu_dai_name = asoc_rtd_to_cpu(rtd, 0)->name; if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -1483,7 +1483,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list) { - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); int paths; if (fe->num_cpus > 1) { @@ -1842,7 +1842,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, * DAIs connected to a single CPU DAI, use CPU DAI's directly */ if (be->num_codecs == 1) { - codec_stream = snd_soc_dai_get_pcm_stream(be->codec_dais[0], stream); + codec_stream = snd_soc_dai_get_pcm_stream(asoc_rtd_to_codec(be, 0), stream); *channels_min = max(*channels_min, codec_stream->channels_min); @@ -2759,7 +2759,7 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) return 0; /* only check active links */ - if (!fe->cpu_dai->active) + if (!asoc_rtd_to_cpu(fe, 0)->active) return 0; /* DAPM sync will call this to update DSP paths */ @@ -2769,13 +2769,13 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) for_each_pcm_streams(stream) { /* skip if FE doesn't have playback/capture capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) || - !snd_soc_dai_stream_valid(fe->codec_dai, stream)) + if (!snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream) || + !snd_soc_dai_stream_valid(asoc_rtd_to_codec(fe, 0), stream)) continue; /* skip if FE isn't currently playing/capturing */ - if (!fe->cpu_dai->stream_active[stream] || - !fe->codec_dai->stream_active[stream]) + if (!asoc_rtd_to_cpu(fe, 0)->stream_active[stream] || + !asoc_rtd_to_codec(fe, 0)->stream_active[stream]) continue; paths = dpcm_path_get(fe, stream, &list); @@ -2922,9 +2922,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) for_each_rtd_codec_dais(rtd, i, codec_dai) { if (rtd->num_cpus == 1) { - cpu_dai = rtd->cpu_dais[0]; + cpu_dai = asoc_rtd_to_cpu(rtd, 0); } else if (rtd->num_cpus == rtd->num_codecs) { - cpu_dai = rtd->cpu_dais[i]; + cpu_dai = asoc_rtd_to_cpu(rtd, i); } else { dev_err(rtd->card->dev, "N cpus to M codecs link is not supported yet\n"); @@ -2971,7 +2971,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, (rtd->num_codecs > 1) ? - "multicodec" : rtd->codec_dai->name, num); + "multicodec" : asoc_rtd_to_codec(rtd, 0)->name, num); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); @@ -3050,8 +3050,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->no_device_suspend = true; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", - (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, - (rtd->num_cpus > 1) ? "multicpu" : rtd->cpu_dai->name); + (rtd->num_codecs > 1) ? "multicodec" : asoc_rtd_to_codec(rtd, 0)->name, + (rtd->num_cpus > 1) ? "multicpu" : asoc_rtd_to_cpu(rtd, 0)->name); return ret; } -- 2.20.1

1 0

Applied "ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime()" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 49648d02008dd75ddd84b16f9abf29f24c383519 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:47:54 +0900 Subject: [PATCH] ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() rtd->cpu_dais/codec_dais are set at soc_new_pcm_runtime(). rtd->num_cpus/codecs should be set there. This patch do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87h7y6bol1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4ad64b96d893..8cef4fd6df36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -490,6 +490,8 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( */ rtd->cpu_dais = &rtd->dais[0]; rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; + rtd->num_cpus = dai_link->num_cpus; + rtd->num_codecs = dai_link->num_codecs; /* * rtd remaining settings @@ -989,7 +991,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, if (!rtd) return -ENOMEM; - rtd->num_cpus = dai_link->num_cpus; for_each_link_cpus(dai_link, i, cpu) { asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu); if (!asoc_rtd_to_cpu(rtd, i)) { @@ -1004,7 +1005,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, rtd->cpu_dai = rtd->cpu_dais[0]; /* Find CODEC from registered CODECs */ - rtd->num_codecs = dai_link->num_codecs; for_each_link_codecs(dai_link, i, codec) { asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec); if (!asoc_rtd_to_codec(rtd, i)) { -- 2.20.1

1 0

Applied "ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 01faf501eefa86fa6f334bb9c71a48498de653e6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:48:14 +0900 Subject: [PATCH] ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups soc_new_pcm_runtime() setups rtd, but code is very random. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87ftdqbokh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8cef4fd6df36..4d617c495864 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -492,17 +492,12 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; rtd->num_cpus = dai_link->num_cpus; rtd->num_codecs = dai_link->num_codecs; - - /* - * rtd remaining settings - */ - rtd->card = card; - rtd->dai_link = dai_link; + rtd->card = card; + rtd->dai_link = dai_link; + rtd->num = card->num_rtd++; /* see for_each_card_rtds */ list_add_tail(&rtd->list, &card->rtd_list); - rtd->num = card->num_rtd; - card->num_rtd++; return rtd; -- 2.20.1

1 0

Applied "ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1729025b04b9f242dca37f50dba8dd3705eb1ea1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:48:27 +0900 Subject: [PATCH] ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais No-one is using cpu_dai/codec_dai/cpu_dais/codec_dais. Let's remove these from snd_soc_pcm_runtime Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87eetabok4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 14 ++++++++------ sound/soc/soc-core.c | 11 +++-------- 2 files changed, 11 insertions(+), 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4dff6745b067..a904086b8ef6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1143,14 +1143,16 @@ struct snd_soc_pcm_runtime { /* runtime devices */ struct snd_pcm *pcm; struct snd_compr *compr; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai **dais; - struct snd_soc_dai **codec_dais; + /* + * dais = cpu_dai + codec_dai + * see + * soc_new_pcm_runtime() + * asoc_rtd_to_cpu() + * asoc_rtd_to_codec() + */ + struct snd_soc_dai **dais; unsigned int num_codecs; - - struct snd_soc_dai **cpu_dais; unsigned int num_cpus; struct delayed_work delayed_work; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d617c495864..7ea9725e33b3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -487,9 +487,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( * dais = [][][][][][][][][][][][][][][][][][] * ^cpu_dais ^codec_dais * |--- num_cpus ---|--- num_codecs --| + * see + * asoc_rtd_to_cpu() + * asoc_rtd_to_codec() */ - rtd->cpu_dais = &rtd->dais[0]; - rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; rtd->num_cpus = dai_link->num_cpus; rtd->num_codecs = dai_link->num_codecs; rtd->card = card; @@ -996,9 +997,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, snd_soc_rtd_add_component(rtd, asoc_rtd_to_cpu(rtd, i)->component); } - /* Single cpu links expect cpu and cpu_dai in runtime data */ - rtd->cpu_dai = rtd->cpu_dais[0]; - /* Find CODEC from registered CODECs */ for_each_link_codecs(dai_link, i, codec) { asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec); @@ -1011,9 +1009,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, snd_soc_rtd_add_component(rtd, asoc_rtd_to_codec(rtd, i)->component); } - /* Single codec links expect codec and codec_dai in runtime data */ - rtd->codec_dai = rtd->codec_dais[0]; - /* Find PLATFORM from registered PLATFORMs */ for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { -- 2.20.1

1 0

[PATCH v1 0/2] Support headset on Tegra boards that use WM8903
by Dmitry Osipenko 14 Apr '20

14 Apr '20

Hello, Some devices have a 4-pin jack instead of a 3-pin and currently the WM8903 configuration is hardcoded to the case of 3-pin jack in the Tegra's ASoC driver. A new device-tree property is required in order to convey that hardware has a 4-pin jack, and thus, microphone's detection needs to be done in a different way. In particular this is needed for Acer A500 tablet device that has a 4-pin headset jack, otherwise userspace sees headset instead of headphones and internal microphone isn't enabled by ALSA UCM rule when it should be. Please review and apply, thanks in advance. Dmitry Osipenko (2): dt-bindings: sound: tegra-wm8903: Document new nvidia,headset property ASoC: tegra: tegra_wm8903: Support nvidia,headset property .../devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + sound/soc/tegra/tegra_wm8903.c | 6 +++++- 2 files changed, 6 insertions(+), 1 deletion(-) -- 2.25.1

4 10

[PATCH 0/4] ASoC: fix some compile warnings
by Jason Yan 14 Apr '20

14 Apr '20

Remove some defined but not used symbols to fix the gcc warnings. Jason Yan (4): ASoC: wm8994: remove wm1811_snd_controls and mixin_boost_tlv ASoC: wm8991: remove defined but not used 'wm8991_dapm_rxvoice_controls' ASoC: wm8990: remove some defined but unused symbols ASoC: wm8900: remove some defined but not used symbols sound/soc/codecs/wm8900.c | 6 --- sound/soc/codecs/wm8990.c | 98 --------------------------------------- sound/soc/codecs/wm8991.c | 8 ---- sound/soc/codecs/wm8994.c | 8 ---- 4 files changed, 120 deletions(-) -- 2.17.2

2 8

[PATCH 1/2] ASoC: intel: soc-acpi-intel-icl-match: remove useless 'rt1308_2_adr'
by Jason Yan 14 Apr '20

14 Apr '20

Fix the following gcc warning: sound/soc/intel/common/soc-acpi-intel-icl-match.c:90:45: warning: ‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=] static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { ^~~~~~~~~~~~ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index ef8500349f2f..16ec9f382b0f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -87,14 +87,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { } }; -static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { - { - .adr = 0x000210025D130800, - .num_endpoints = 1, - .endpoints = &single_endpoint, - } -}; - static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { { .adr = 0x000110025D130800, -- 2.17.2

3 4

[PATCH v3 1/3] dt-bindings: sound: convert rockchip spdif bindings to yaml
by Johan Jonker 14 Apr '20

14 Apr '20

Current dts files with 'spdif' nodes are manually verified. In order to automate this process rockchip-spdif.txt has to be converted to yaml. Also rk3188.dtsi, rk3288.dtsi use an extra fallback string, so change this in the documentation. Changed: "rockchip,rk3188-spdif", "rockchip,rk3066-spdif" "rockchip,rk3288-spdif", "rockchip,rk3066-spdif" Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes v3: Add reviewed by Fix irq.h already included in arm-gic.h Change rk3188-cru-common.h to rk3188-cru.h Changed v2: dmas and dma-names layout --- .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ----------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 93 ++++++++++++++++++++++ 2 files changed, 93 insertions(+), 45 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt deleted file mode 100644 index ec20c1271..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Rockchip SPDIF transceiver - -The S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-spdif" - - "rockchip,rk3188-spdif" - - "rockchip,rk3228-spdif" - - "rockchip,rk3288-spdif" - - "rockchip,rk3328-spdif" - - "rockchip,rk3366-spdif" - - "rockchip,rk3368-spdif" - - "rockchip,rk3399-spdif" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the SPDIF interrupt. -- dmas: DMA specifiers for tx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should be "tx" -- clocks: a list of phandle + clock-specifier pairs, one for each entry - in clock-names. -- clock-names: should contain following: - - "hclk": clock for SPDIF controller - - "mclk" : clock for SPDIF bus - -Required properties on RK3288: - - rockchip,grf: the phandle of the syscon node for the general register - file (GRF) - -Example for the rk3188 SPDIF controller: - -spdif: spdif@1011e000 { - compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; - reg = <0x1011e000 0x2000>; - interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dmac1_s 8>; - dma-names = "tx"; - clock-names = "hclk", "mclk"; - clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..bfd44fd8e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + maxItems: 1 + + dma-names: + const: tx + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + }; -- 2.11.0

2 5

Applied "ASoC: rockchip-spdif: add #sound-dai-cells property" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: rockchip-spdif: add #sound-dai-cells property has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f0c0f5a4541cb5dee1f16cebcc4e11d952394c53 Mon Sep 17 00:00:00 2001 From: Johan Jonker <jbx6244(a)gmail.com> Date: Sat, 4 Apr 2020 13:52:24 +0200 Subject: [PATCH] ASoC: rockchip-spdif: add #sound-dai-cells property '#sound-dai-cells' is required to properly interpret the list of DAI specified in the 'sound-dai' property, so add them to 'rockchip-spdif.yaml' Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20200404115225.4314-2-jbx6244@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index bfd44fd8ecbc..f381dbbf5de6 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -57,6 +57,9 @@ properties: The phandle of the syscon node for the GRF register. Required property on RK3288. + "#sound-dai-cells": + const: 0 + required: - compatible - reg @@ -65,6 +68,7 @@ required: - clock-names - dmas - dma-names + - "#sound-dai-cells" if: properties: @@ -90,4 +94,5 @@ examples: clock-names = "mclk", "hclk"; dmas = <&dmac1_s 8>; dma-names = "tx"; + #sound-dai-cells = <0>; }; -- 2.20.1

1 0

Applied "ASoC: rockchip-spdif: add power-domains property" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: rockchip-spdif: add power-domains property has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 65f1d3d9cee7e10d105756b6ef678885a23a31d7 Mon Sep 17 00:00:00 2001 From: Johan Jonker <jbx6244(a)gmail.com> Date: Sat, 4 Apr 2020 13:52:25 +0200 Subject: [PATCH] ASoC: rockchip-spdif: add power-domains property In the old txt situation we add/describe only properties that are used by the driver/hardware itself. With yaml it also filters things in a node that are used by other drivers like 'power-domains' for rk3399, so add it to 'rockchip-spdif.yaml'. Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20200404115225.4314-3-jbx6244@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index f381dbbf5de6..c467152656f7 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -51,6 +51,9 @@ properties: dma-names: const: tx + power-domains: + maxItems: 1 + rockchip,grf: $ref: /schemas/types.yaml#/definitions/phandle description: -- 2.20.1

1 0

[PATCH 1/3] ASoC: dapm: Fix regression introducing multiple copies of DAI widgets
by Charles Keepax 14 Apr '20

14 Apr '20

Refactoring was done to factor out the linking of DAI widgets into a helper function, dapm_add_valid_dai_widget. However when this was done, a regression was introduced for CODEC to CODEC links. It was over looked that the playback and capture variables persisted across all CODEC DAIs being processed, which ensured that the special DAI widget that is added for CODEC to CODEC links was only created once. This bug causes kernel panics during DAPM shutdown. To stick with the spirit of the original refactoring whilst fixing the issue, variables to hold the DAI widgets are added to snd_soc_dai_link. Furthermore the dapm_add_valid_dai_widget function is renamed to dapm_connect_dai_pair, the function only adds DAI widgets in the CODEC to CODEC case and its primary job is to add routes connecting two DAI widgets, making the original name quite misleading. Fixes: 6c4b13b51aa3 ("ASoC: Add dapm_add_valid_dai_widget helper") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc.h | 3 ++ sound/soc/soc-dapm.c | 91 +++++++++++++++++++++++++++------------------------- 2 files changed, 51 insertions(+), 43 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 13458e4fbb132..946f88a6c63d1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -790,6 +790,9 @@ struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; + struct snd_soc_dapm_widget *playback_widget; + struct snd_soc_dapm_widget *capture_widget; + unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 679ed60d850ec..fe907f0cc709b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4283,52 +4283,63 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_add_valid_dai_widget(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai) +static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm, + struct snd_soc_dai *src_dai, + struct snd_soc_dapm_widget *src, + struct snd_soc_dapm_widget *dai, + struct snd_soc_dai *sink_dai, + struct snd_soc_dapm_widget *sink) { - struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; - struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; + dev_dbg(dapm->dev, "connected DAI link %s:%s -> %s:%s\n", + src_dai->component->name, src->name, + sink_dai->component->name, sink->name); + + if (dai) { + snd_soc_dapm_add_path(dapm, src, dai, NULL, NULL); + src = dai; + } + + snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL); +} + +static void dapm_connect_dai_pair(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_dapm_widget *dai, *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - if (rtd->dai_link->params) { + if (dai_link->params) { playback_cpu = cpu_dai->capture_widget; capture_cpu = cpu_dai->playback_widget; } else { - playback = cpu_dai->playback_widget; - capture = cpu_dai->capture_widget; - playback_cpu = playback; - capture_cpu = capture; + playback_cpu = cpu_dai->playback_widget; + capture_cpu = cpu_dai->capture_widget; } /* connect BE DAI playback if widgets are valid */ codec = codec_dai->playback_widget; if (playback_cpu && codec) { - if (!playback) { + if (dai_link->params && !dai_link->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { + dai = snd_soc_dapm_new_dai(card, substream, "playback"); + if (IS_ERR(dai)) { dev_err(rtd->dev, "ASoC: Failed to create DAI %s: %ld\n", codec_dai->name, - PTR_ERR(playback)); + PTR_ERR(dai)); goto capture; } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + dai_link->playback_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu, + dai_link->playback_widget, + codec_dai, codec); } capture: @@ -4336,28 +4347,22 @@ static void dapm_add_valid_dai_widget(struct snd_soc_card *card, codec = codec_dai->capture_widget; if (codec && capture_cpu) { - if (!capture) { + if (dai_link->params && !dai_link->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { + dai = snd_soc_dapm_new_dai(card, substream, "capture"); + if (IS_ERR(dai)) { dev_err(rtd->dev, "ASoC: Failed to create DAI %s: %ld\n", codec_dai->name, - PTR_ERR(capture)); + PTR_ERR(dai)); return; } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dai_link->capture_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); + dapm_connect_dai_routes(&card->dapm, codec_dai, codec, + dai_link->capture_widget, + cpu_dai, capture_cpu); } } @@ -4369,12 +4374,12 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, if (rtd->num_cpus == 1) { for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[0]); + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[0]); } else if (rtd->num_codecs == rtd->num_cpus) { for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[i]); + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[i]); } else { dev_err(card->dev, "N cpus to M codecs link is not supported yet\n"); -- 2.11.0

2 5

[PATCH] soc/stm/stm32_sub_sai: Add missing '\n' in log messages
by Sebastian Fricke 14 Apr '20

14 Apr '20

Message logged by 'dev_xxx()' or 'pr_xxx()' should end with a '\n'. Fixes: 3e086ed("ASoC: stm32: add SAI drivers") Signed-off-by: Sebastian Fricke <sebastian.fricke.linux(a)gmail.com> --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 0d0c9afd8791..34a7c3d6fb91 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -837,7 +837,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_32); break; default: - dev_err(cpu_dai->dev, "Data format not supported"); + dev_err(cpu_dai->dev, "Data format not supported\n"); return -EINVAL; } -- 2.20.1

2 1

[PATCH] ASoC: qcom: common: Silence duplicate parse error messages
by Bjorn Andersson 14 Apr '20

14 Apr '20

All error paths in qcom_snd_parse_of() prints more specific error messages, so silence the one in apq8096_platform_probe() and sdm845_snd_platform_probe() to avoid spamming the kernel log. Signed-off-by: Bjorn Andersson <bjorn.andersson(a)linaro.org> --- sound/soc/qcom/apq8096.c | 4 +--- sound/soc/qcom/sdm845.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index d55e3ad96716..287ad2aa27f3 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -116,10 +116,8 @@ static int apq8096_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto err; - } apq8096_add_be_ops(card); ret = snd_soc_register_card(card); diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index b2de65c7f95c..68e9388ff46f 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -559,10 +559,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto parse_dt_fail; - } data->card = card; snd_soc_card_set_drvdata(card, data); -- 2.24.0

2 1

Applied "ASoC: wsa881x: mark read_only_wordlength flag" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: wsa881x: mark read_only_wordlength flag has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0f2a3b02274c02eb97697c4d89c019d1d21ac225 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Tue, 14 Apr 2020 12:03:47 +0100 Subject: [PATCH] ASoC: wsa881x: mark read_only_wordlength flag WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wsa881x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f2d6f2f81f14..d39d479e2378 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* COMP */ .num = 2, @@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* BOOST */ .num = 3, @@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* VISENSE */ .num = 4, @@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, } }; -- 2.20.1

1 0

[PATCH] ASoC: madera: Remove a couple of stray blank lines
by Charles Keepax 14 Apr '20

14 Apr '20

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/madera.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 40de9d7811d1f..a448d2a2918a9 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -1903,7 +1903,6 @@ const struct soc_enum madera_isrc_fsh[] = { MADERA_ISRC4_FSH_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsh); @@ -1924,7 +1923,6 @@ const struct soc_enum madera_isrc_fsl[] = { MADERA_ISRC4_FSL_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsl); @@ -1938,7 +1936,6 @@ const struct soc_enum madera_asrc1_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc1_rate); @@ -1964,7 +1961,6 @@ const struct soc_enum madera_asrc2_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc2_rate); -- 2.11.0

2 1

[PATCH v2 1/3] dt-bindings: sound: convert rockchip i2s bindings to yaml
by Johan Jonker 14 Apr '20

14 Apr '20

Current dts files with 'i2s' nodes are manually verified. In order to automate this process rockchip-i2s.txt has to be converted to yaml. Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changed V2: Add Reviewed-by --- .../devicetree/bindings/sound/rockchip-i2s.txt | 49 ---------- .../devicetree/bindings/sound/rockchip-i2s.yaml | 106 +++++++++++++++++++++ 2 files changed, 106 insertions(+), 49 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt deleted file mode 100644 index 54aefab71..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ /dev/null @@ -1,49 +0,0 @@ -* Rockchip I2S controller - -The I2S bus (Inter-IC sound bus) is a serial link for digital -audio data transfer between devices in the system. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-i2s": for rk3066 - - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 - - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 - - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 - - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 - - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the I2S interrupt. -- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should contain the following: - - "i2s_hclk": clock for I2S BUS - - "i2s_clk" : clock for I2S controller -- rockchip,playback-channels: max playback channels, if not set, 8 channels default. -- rockchip,capture-channels: max capture channels, if not set, 2 channels default. - -Required properties for controller which support multi channels -playback/capture: - -- rockchip,grf: the phandle of the syscon node for GRF register. - -Example for rk3288 I2S controller: - -i2s@ff890000 { - compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; - reg = <0xff890000 0x10000>; - interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&pdma1 0>, <&pdma1 1>; - dma-names = "tx", "rx"; - clock-names = "i2s_hclk", "i2s_clk"; - clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; - rockchip,playback-channels = <8>; - rockchip,capture-channels = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 000000000..eff06b4b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,106 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S controller + +description: + The I2S bus (Inter-IC sound bus) is a serial link for digital + audio data transfer between devices in the system. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-i2s + - items: + - enum: + - rockchip,px30-i2s + - rockchip,rk3036-i2s + - rockchip,rk3188-i2s + - rockchip,rk3228-i2s + - rockchip,rk3288-i2s + - rockchip,rk3328-i2s + - rockchip,rk3366-i2s + - rockchip,rk3368-i2s + - rockchip,rk3399-i2s + - const: rockchip,rk3066-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for I2S controller + - description: clock for I2S BUS + + clock-names: + items: + - const: i2s_clk + - const: i2s_hclk + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + rockchip,capture-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + description: + Max capture channels, if not set, 2 channels default. + + rockchip,playback-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 8 + description: + Max playback channels, if not set, 8 channels default. + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property for controllers which support multi channel + playback/capture. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3288-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>; + clock-names = "i2s_clk", "i2s_hclk"; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + rockchip,capture-channels = <2>; + rockchip,playback-channels = <8>; + }; -- 2.11.0

3 4

Regression on Intel Atom/SST/Baytrail devices w/ v5.7-rc1
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

Hi, With Mark's for-next tree and v5.7-rc1 I get these really bad errors with the Atom/SST driver on a Baytrail device - found by accident while trying to validate UCM changes. [ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed [ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed Trying to bisect, the errors change at some point: [c77b8317ee3ab43634421afb73fdb1ea253d3d47] ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out [ 9.387540] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for Headset Playback [ 9.387551] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for media1_in [ 9.387556] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for media0_out mix 0 [ 9.387561] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for media0_out [ 9.387566] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for pcm0_in [ 9.387570] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for modem_out mix 0 [ 9.387575] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for modem_out [ 9.387580] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for codec_out0 mix 0 [ 9.387584] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for codec_out0 And then from [5d3c644773925c3568617435e42a9404a114c428] drm/mediatek: support HDMI jack status reporting to v5.6-rc1 I get those time outs... [ 21.031638] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 21.031661] intel_sst_acpi 80860F28:00: fw returned err -16 [ 21.031692] sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16 Both 5.5 and 5.6 work though, as if we are missing a set of dependencies? I am running out of time this evening, but I'd be good to see if anyone else reproduces my findings. Thanks -Pierre

2 2

[alsa-devel] [PATCH] alsactl: avoid needless wakeups in monitor loop.
by Zev Weiss 14 Apr '20

14 Apr '20

The timeout wasn't really being used for anything; disabling it should reduce idle energy consumption slightly. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..6d9328d899a4 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -340,7 +340,7 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, int count; int i; - count = epoll_wait(epfd, epev, max_ev_count, 200); + count = epoll_wait(epfd, epev, max_ev_count, -1); if (count < 0) { err = count; break; -- 2.23.0

3 3

[alsa-devel] [PATCH] alsactl: don't exit on EINTR from epoll_wait().
by Zev Weiss 14 Apr '20

14 Apr '20

Previously, things like attaching strace to a running 'alsactl monitor' process would cause it to exit. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..cf4167bee165 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -342,6 +342,8 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, count = epoll_wait(epfd, epev, max_ev_count, 200); if (count < 0) { + if (errno == EINTR) + continue; err = count; break; } -- 2.23.0

3 2

[PATCH] ASoC: tas571x: disable regulators on failed probe
by Philipp Puschmann 14 Apr '20

14 Apr '20

If probe fails after enabling the regulators regulator_put is called for each supply without having them disabled before. This produces some warnings like WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c [<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14) [<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4) [<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4) [<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c) [<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38) [<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38) [<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c) [<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c) [<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c) [<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60) [<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc) [<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0) [<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0) [<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108) [<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88) [<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250) [<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244) [<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540) [<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8) [<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20) Fixes: 3fd6e7d9a146 (ASoC: tas571x: New driver for TI TAS571x power amplifiers) Signed-off-by: Philipp Puschmann <p.puschmann(a)pironex.de> --- sound/soc/codecs/tas571x.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 1554631cb397..5b7f9fcf6cbf 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, priv->regmap = devm_regmap_init(dev, NULL, client, priv->chip->regmap_config); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + goto disable_regs; + } priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW); if (IS_ERR(priv->pdn_gpio)) { @@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0); if (ret) - return ret; + goto disable_regs; usleep_range(50000, 60000); @@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client, */ ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0); if (ret) - return ret; + goto disable_regs; } - return devm_snd_soc_register_component(&client->dev, + ret = devm_snd_soc_register_component(&client->dev, &priv->component_driver, &tas571x_dai, 1); + if (ret) + goto disable_regs; + + return ret; + +disable_regs: + regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies); + return ret; } static int tas571x_i2c_remove(struct i2c_client *client) -- 2.26.0

2 2

Applied "ASoC: tas571x: disable regulators on failed probe" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: tas571x: disable regulators on failed probe has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9df8ba7c63073508e5aa677dade48fcab6a6773e Mon Sep 17 00:00:00 2001 From: Philipp Puschmann <p.puschmann(a)pironex.de> Date: Tue, 14 Apr 2020 13:27:54 +0200 Subject: [PATCH] ASoC: tas571x: disable regulators on failed probe If probe fails after enabling the regulators regulator_put is called for each supply without having them disabled before. This produces some warnings like WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c [<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14) [<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4) [<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4) [<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c) [<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38) [<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38) [<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c) [<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c) [<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c) [<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60) [<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc) [<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0) [<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0) [<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108) [<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88) [<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250) [<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244) [<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540) [<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8) [<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20) Fixes: 3fd6e7d9a146 (ASoC: tas571x: New driver for TI TAS571x power amplifiers) Signed-off-by: Philipp Puschmann <p.puschmann(a)pironex.de> Link: https://lore.kernel.org/r/20200414112754.3365406-1-p.puschmann@pironex.de Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas571x.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 1554631cb397..5b7f9fcf6cbf 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, priv->regmap = devm_regmap_init(dev, NULL, client, priv->chip->regmap_config); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + goto disable_regs; + } priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW); if (IS_ERR(priv->pdn_gpio)) { @@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0); if (ret) - return ret; + goto disable_regs; usleep_range(50000, 60000); @@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client, */ ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0); if (ret) - return ret; + goto disable_regs; } - return devm_snd_soc_register_component(&client->dev, + ret = devm_snd_soc_register_component(&client->dev, &priv->component_driver, &tas571x_dai, 1); + if (ret) + goto disable_regs; + + return ret; + +disable_regs: + regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies); + return ret; } static int tas571x_i2c_remove(struct i2c_client *client) -- 2.20.1

1 0

[PATCH] ASoC: amd: Fix button configuration
by Akshu Agrawal 14 Apr '20

14 Apr '20

RT5682 buttons were incorrectly mapped. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 024a7ee54cd5..e499c00e0c66 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -89,9 +89,9 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) } snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ret = snd_soc_component_set_jack(component, &pco_jack, NULL); if (ret) { -- 2.20.1

2 1

[RESEND PATCH] ASoC: wsa881x: mark read_only_wordlength flag
by Srinivas Kandagatla 14 Apr '20

14 Apr '20

WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi Mark, For some reason this patch was missed during last merge window, Other patch in this series is already in mainline. Without this patch audio is not functional on DB845c and other SDM845 based platforms. Can you please take this for next possible rc. Thanks, srini sound/soc/codecs/wsa881x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f2d6f2f81f14..d39d479e2378 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* COMP */ .num = 2, @@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* BOOST */ .num = 3, @@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* VISENSE */ .num = 4, @@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, } }; -- 2.21.0

2 1

[PATCH] ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume
by Krzysztof Kozlowski 14 Apr '20

14 Apr '20

Commit 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") removed the DAI side suspend/resume hooks and switched entirely to component suspend/resume. However the Samsung SoC s3c-i2s-v2 driver was not updated. Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while changing dai to component which allows to keep the struct snd_soc_component_driver const. This fixes build errors: sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’: sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’ dai_drv->suspend = s3c2412_i2s_suspend; Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Only compile tested. --- sound/soc/samsung/s3c-i2s-v2.c | 57 --------------------------------- sound/soc/samsung/s3c2412-i2s.c | 56 ++++++++++++++++++++++++++++++++ 2 files changed, 56 insertions(+), 57 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 358887848293..5e95c30fb2ba 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup); -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warn("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warn("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warn("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - int s3c_i2sv2_register_component(struct device *dev, int id, const struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) @@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id, if (!ops->delay) ops->delay = s3c2412_i2s_delay; - dai_drv->suspend = s3c2412_i2s_suspend; - dai_drv->resume = s3c2412_i2s_resume; - return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..b35d828c1cfe 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + u32 iismod; + + if (component->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warn("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warn("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warn("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + + pr_info("component_active %d, IISMOD %08x, IISCON %08x\n", + component->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (component->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static const struct snd_soc_component_driver s3c2412_i2s_component = { .name = "s3c2412-i2s", + .suspend = s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, }; static int s3c2412_iis_dev_probe(struct platform_device *pdev) -- 2.17.1

2 1

[RFC v2] ALSA: control: fix a error handling exist in snd_ctl_elem_add
by yangerkun 14 Apr '20

14 Apr '20

CVE-2020-11725 report that 'count = info->owner' may result a SIZE_OVERFLOW. 'info->owner' represent a pid, and actually, we should use info->count. Signed-off-by: yangerkun <yangerkun(a)huawei.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) v1->v2: reword the patch head diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..c77ca7f39637 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1431,7 +1431,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return -ENOMEM; /* Check the number of elements for this userspace control. */ - count = info->owner; + count = info->count; if (count == 0) count = 1; -- 2.21.1

2 2

[RFC] ALSA: control: fix a mishandled exits in snd_ctl_elem_add
by yangerkun 14 Apr '20

14 Apr '20

CVE-2020-11725 report that 'count = info->owner' may result a SIZE_OVERFLOW. 'info->owner' represent a pid, and actually, we should use info->count. Signed-off-by: yangerkun <yangerkun(a)huawei.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..c77ca7f39637 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1431,7 +1431,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return -ENOMEM; /* Check the number of elements for this userspace control. */ - count = info->owner; + count = info->count; if (count == 0) count = 1; -- 2.21.1

1 0

[PATCH v7 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 14 Apr '20

14 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 v7 - updated according to Nicoin's comments. - add get_pair_priv_size to replace PAIR_PRIVATE_SIZE 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 | 24 +- sound/soc/fsl/fsl_asrc.c | 310 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2119 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3222 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

3 11

[PATCH] ALSA: hda: Allow setting preallocation again for x86
by Takashi Iwai 13 Apr '20

13 Apr '20

The commit c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default to zero for x86, as the preallocation should work almost all cases. However, this expectation was too naive; some applications try to allocate as the max buffer size as possible, and it leads to the memory exhaustion. More badly, the commit changed the kconfig no longer adjustable for x86, so you can't fix it statically (although it can be still adjusted via procfs). So, practically seen, it's more recommended to set a reasonable limit for x86, too. This patch follows to that experience, and changes the default to 2048 and allow the kconfig adjustable again. Fixes: c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/Kconfig | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 4ca6b09056f3..3bc9224d5e4f 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -21,16 +21,17 @@ config SND_HDA_EXT_CORE select SND_HDA_CORE config SND_HDA_PREALLOC_SIZE - int "Pre-allocated buffer size for HD-audio driver" if !SND_DMA_SGBUF + int "Pre-allocated buffer size for HD-audio driver" range 0 32768 - default 0 if SND_DMA_SGBUF + default 2048 if SND_DMA_SGBUF default 64 if !SND_DMA_SGBUF help Specifies the default pre-allocated buffer-size in kB for the HD-audio driver. A larger buffer (e.g. 2048) is preferred for systems using PulseAudio. The default 64 is chosen just for compatibility reasons. - On x86 systems, the default is zero as we need no preallocation. + On x86 systems, the default is 2048 as a reasonable value for + most of modern systems. Note that the pre-allocation size can be changed dynamically via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. -- 2.16.4

1 0

SOF support for Baytrail/CherryTrail
by GitHub pull_request - edited 13 Apr '20

13 Apr '20

alsa-project/alsa-ucm-conf pull request #20 was edited from plbossart: Add symlink and simple test to reuse legacy settings as is. tested on Baytrail/Asus T100TA only with SOF and legacy drivers, all other platforms are just based on same changes. sof-bytcr-rt5640 is here: http://alsa-project.org/db/?f=6f8be0520e2bbda94cb744ce66232174feec03d4 UCM validator seems to have problems with the conditional statement? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fix/byt
by GitHub pull_request - opened 13 Apr '20

13 Apr '20

alsa-project/alsa-ucm-conf pull request #20 was opened from plbossart: Add symlink and simple test to reuse legacy settings as is. tested on Baytrail/Asus T100TA only with SOF and legacy drivers, all other platforms are just based on same changes. sof-bytcr-rt5640 is here: http://alsa-project.org/db/?f=6f8be0520e2bbda94cb744ce66232174feec03d4 UCM validator seems to have problems with the conditional statement? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/5] bytcr-rt5651: Fix dmic check in HiFi-Components.conf
by Hans de Goede 13 Apr '20

13 Apr '20

The kernel sets cfg-mic:dmic not cfg-mic:dmic1 when a digital mic is used. The wrong check for cfg-mic:dmic1 causes there to be no "Mic" device at all, which causes the check for the ConflictingDevices section of the HeadSet (mic) device to fail like e.g. so: [hans@localhost ~]$ alsaucm -c bytcr-rt5651 set _verb HiFi set _enadev Speaker ALSA lib parser.c:1138:(verb_dev_list_add) error: unable to find device 'Mic' ALSA lib parser.c:1433:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:962:(snd_use_case_mgr_open) error: failed to import bytcr-rt5651 use case configuration -2 alsaucm: error failed to open sound card bytcr-rt5651: No such file or directory This commit fixes this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/bytcr-rt5651/HiFi-Components.conf | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/ucm2/bytcr-rt5651/HiFi-Components.conf b/ucm2/bytcr-rt5651/HiFi-Components.conf index 53ebcbe..d0a573e 100644 --- a/ucm2/bytcr-rt5651/HiFi-Components.conf +++ b/ucm2/bytcr-rt5651/HiFi-Components.conf @@ -34,11 +34,11 @@ If.headphones { } } -If.dmic1 { +If.dmic { Condition { Type String Haystack "${CardComponents}" - Needle "cfg-mic:dmic1" + Needle "cfg-mic:dmic" } True { <codecs/rt5651/DigitalMic.conf> -- 2.26.0

5 24

[PATCH 0/6] ALSA: hda: Fix/improve no-codec bus
by Takashi Iwai 13 Apr '20

13 Apr '20

Hi, here is a patch set for fixing / improving the error handling of HD-audio bus without codecs, which can be seen on some GPUs, for example. It's been debugged and tested in bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Takashi === Roy Spliet (1): ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported Takashi Iwai (5): ALSA: hda: Don't release card at firmware loading error ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq ops ALSA: hda: Release resources at error in delayed probe ALSA: hda: Keep the controller initialization even if no codecs found ALSA: hda: Skip controller resume if not needed include/sound/hda_codec.h | 5 +++ sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 106 +++++++++++++++++++++++++++------------------- sound/pci/hda/hda_intel.h | 1 + 4 files changed, 69 insertions(+), 45 deletions(-) -- 2.16.4

2 7

[PATCH] ASoC: bdw-rt5650: incorrect rate of PCM data
by Brent Lu 13 Apr '20

13 Apr '20

Implement a constrain to exclude 3-channel capture since only 2 and 4 channel capture are supported on the platform. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5650.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index af2f502..eedbdad 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -83,6 +83,36 @@ static struct snd_soc_jack_pin mic_jack_pin = { .mask = SND_JACK_MICROPHONE, }; +static const unsigned int channels[] = { + 2, 4, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int bdw_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 2 or 4 channel capture + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return 0; +} + +static const struct snd_soc_ops bdw_rt5650_fe_ops = { + .startup = bdw_fe_startup, +}; + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -234,6 +264,7 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { .name = "System PCM", .stream_name = "System Playback", .dynamic = 1, + .ops = &bdw_rt5650_fe_ops, #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) .init = bdw_rt5650_rtd_init, #endif -- 2.7.4

3 2

alsa-gobject: seq: tstamp: use wrapper structure instead of union
by GitHub pull_request - opened 13 Apr '20

13 Apr '20

alsa-project/alsa-gobject pull request #17 was opened from takaswie: In UAPI for ALSA Sequencer, snd_seq_timestamp is union with snd_seq_tick_time_t and struct snd_seq_real_time types. The libalsaseq has GLib Boxed object for the union, named as ALSASeqTstamp. ``` $ cat build/src/seq/ALSASeq-0.0.gir ... <union name="Tstamp" c:type="ALSASeqTstamp" glib:type-name="ALSASeqTstamp" glib:get-type="alsaseq_tstamp_get_type" c:symbol-prefix="tstamp"> <source-position filename="../src/seq/tstamp.h" line="14"/> ... </union> ... ``` Although this is valid in a view of GObject Introspection, it's not necessarily convenient to language bindings because some programming language doesn't support union type. In this case, the type is not available. This commit adds an alternative structure to wrap the union. As a result, ALSASeqTstamp is GLib Boxed object for the wrapper structure. ``` $ cat build/src/seq/ALSASeq-0.0.gir ... <record name="Tstamp" c:type="ALSASeqTstamp" glib:type-name="ALSASeqTstamp" glib:get-type="alsaseq_tstamp_get_type" c:symbol-prefix="tstamp"> <source-position filename="../src/seq/tstamp.h" line="18"/> <field name="tstamp" writable="1"> <type name="gpointer" c:type="snd_seq_timestamp"/> </field> ... </record> ... ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/17 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/17.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH RFC] clumsy attempt at fixing rate plugin draining function
by sylvain.bertrand＠gmail.com 12 Apr '20

12 Apr '20

Hi, This is related to github issue #41. I don't know how to update last_commit_ptr properly. Neither if it is consistent with the rate state (boundary, app_ptr, etc). -- Sylvain --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1051,6 +1051,7 @@ static int snd_pcm_rate_drain(snd_pcm_t *pcm) /* commit the remaining fraction (if any) */ snd_pcm_uframes_t size, ofs, saved_avail_min; snd_pcm_sw_params_t sw_params; + int commit_err; __snd_pcm_lock(pcm); /* temporarily set avail_min to one */ @@ -1064,7 +1065,7 @@ static int snd_pcm_rate_drain(snd_pcm_t *pcm) while (size > 0) { snd_pcm_uframes_t psize, spsize; int err; - + err = __snd_pcm_wait_in_lock(rate->gen.slave, -1); if (err < 0) break; @@ -1077,14 +1078,29 @@ static int snd_pcm_rate_drain(snd_pcm_t *pcm) if (! spsize) break; } - snd_pcm_rate_commit_area(pcm, rate, ofs, + commit_err = snd_pcm_rate_commit_area(pcm, rate, ofs, psize, spsize); + if (commit_err == 1) { + rate->last_commit_ptr += psize; + if (rate->last_commit_ptr >= pcm->boundary) + rate->last_commit_ptr = 0; + } else if (commit_err == 0) { + if (pcm->mode & SND_PCM_NONBLOCK != 0) { + commit_err = -EAGAIN; + break; + } + continue; + } else + break; + ofs = (ofs + psize) % pcm->buffer_size; size -= psize; } sw_params.avail_min = saved_avail_min; snd_pcm_sw_params(rate->gen.slave, &sw_params); __snd_pcm_unlock(pcm); + if (commit_err < 0) + return commit_err; } return snd_pcm_drain(rate->gen.slave); }

1 0

desperate to find a fix for ALC298
by n c 12 Apr '20

12 Apr '20

Kernel: linux5.6 (source here <https://github.com/torvalds/linux/releases/tag/v5.6>) Distro: Tried most for months since linux5.4 (currently on manjaro) Desktop: gnome Alsa-info: (pastebin here <https://pastebin.com/w35JGdGu>) Any configs: No just the defaults (no alsa/.asound, pulse/client.conf etc...) Cards: Single on-board Realtek ALC298 *Issue* On-board sound does not work at all. Neither on speakers or headphones, no matter which distro or kernel I tried (since 5.4+ of course). *Have I asked the community?* I have a SO question here <https://unix.stackexchange.com/questions/579486/alsa-pulseaudio-and-intel-h…> and a manjaro topic here <https://forum.manjaro.org/t/alsa-pulseaudio-and-intel-hda-pch-with-no-sound> . *Have I tried fixing it myself?* Yes, a lot. The threads above both have a lot of information in them. Highlight of my debugging so far is that the codec pin "0x1a" has an incorrect VREF assigned to it (HIZ). Found this by accident when using hda-analyzer. Changing it to VREF 50, 80 or 100 fixes the headphones. Have no idea how to persist it using init_verbs (don't know how to package the new value 0x05), so wrote a systemd file to hack it using hda-verb utility at startup. It stopped working eventually or sometimes had to stop/restart the service to reissue the change. I read somewhere that the incorrect VREF is a sign of UEFI/BIOS chip assigning incorrect speaker pins. Although I could not get the speakers working at all using any utilities mentioned above. Thank you for reading,

1 0

[PATCH 0/4] ALSA: usb-audio: More workarounds for TRX40-mobos
by Takashi Iwai 12 Apr '20

12 Apr '20

Hi, it turned out that the recent mobos shipped with Realtek codec need more workarounds. In the end, most of the fixes here are generic issues, hence marked for stable. The bugtracker is at: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Takashi === Takashi Iwai (4): ALSA: usb-audio: Filter error from connector kctl ops, too ALSA: usb-audio: Don't override ignore_ctl_error value from the map ALSA: usb-audio: Don't create jack controls for PCM terminals ALSA: usb-audio: Check mapping at creating connector controls, too sound/usb/mixer.c | 31 +++++++++++++++++++------------ sound/usb/mixer_maps.c | 4 +++- 2 files changed, 22 insertions(+), 13 deletions(-) -- 2.16.4

1 4

snd_pcm_drain() infinite loop
by GitHub issues - edited 12 Apr '20

12 Apr '20

alsa-project/alsa-lib issue #41 was edited from sylware: In a non-blocking dmix-ed plugin pipeline (see below), playing audio from a 44100Hz source to a running 48000Hz hardware pcm, will go into an -EAGAIN infinite loop in snd_pcm_drain(). I did a bit of tracing and it seems snd_pcm_drain() will always return -EAGAIN because in pcm_dmix.c:snd_pcm_dmix_sync_ptr0, avail is never >= pcm->stop_threshold Issue URL : https://github.com/alsa-project/alsa-lib/issues/41 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 4.9 22/26] ASoC: Intel: Skylake: Enable codec wakeup during chip init
by Sasha Levin 12 Apr '20

12 Apr '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e603f11d5df8997d104ab405ff27640b90baffaa ] Follow the recommendation set by hda_intel.c and enable HDMI/DP codec wakeup during bus initialization procedure. Disable wakeup once init completes. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 3e526fbd267e6..223958c49bf8d 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -102,9 +102,11 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) { int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 31/37] ASoC: Intel: Skylake: Enable codec wakeup during chip init
by Sasha Levin 12 Apr '20

12 Apr '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e603f11d5df8997d104ab405ff27640b90baffaa ] Follow the recommendation set by hda_intel.c and enable HDMI/DP codec wakeup during bus initialization procedure. Disable wakeup once init completes. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index a0bef63b8fb11..7fb4df78ecb35 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -103,9 +103,11 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) { int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 53/66] ASoC: Intel: Skylake: Enable codec wakeup during chip init
by Sasha Levin 12 Apr '20

12 Apr '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e603f11d5df8997d104ab405ff27640b90baffaa ] Follow the recommendation set by hda_intel.c and enable HDMI/DP codec wakeup during bus initialization procedure. Disable wakeup once init completes. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 50f16a0f6535f..463078c2346f9 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -128,6 +128,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) struct hdac_ext_link *hlink; int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); @@ -136,6 +137,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 086/108] ASoC: Intel: Skylake: Enable codec wakeup during chip init
by Sasha Levin 12 Apr '20

12 Apr '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e603f11d5df8997d104ab405ff27640b90baffaa ] Follow the recommendation set by hda_intel.c and enable HDMI/DP codec wakeup during bus initialization procedure. Disable wakeup once init completes. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975acd..9ef9651c4dfe1 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -129,6 +129,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) struct hdac_ext_link *hlink; int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); @@ -137,6 +138,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 085/108] ASoC: stm32: spdifrx: fix regmap status check
by Sasha Levin 12 Apr '20

12 Apr '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit a168dae5ea14283e8992d5282237bb0d6a3e1c06 ] Release resources when exiting on error. Fixes: 1a5c0b28fc56 ("ASoC: stm32: spdifrx: manage identification registers") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Link: https://lore.kernel.org/r/20200318144125.9163-2-olivier.moysan@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_spdifrx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index e53fb4bd66b36..9fc2a1767eb1d 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -995,6 +995,8 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) if (idr == SPDIFRX_IPIDR_NUMBER) { ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver); + if (ret) + goto error; dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n", FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver), -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 058/108] ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed
by Sasha Levin 12 Apr '20

12 Apr '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit 8726ee6148fe24e2b29d4a961ad95c4ff8025d1d ] 1. Switch TDM GPIO pins according to playback on or off. 2. Pull TDM GPIO pins down when probed to avoid current leakage. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.2.I1d568b0c99742c6e755d051aadfd52e… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 104 +++++++++++++++++- 1 file changed, 98 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 43f99e59a0786..897a7bc4d3f3f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,6 +18,22 @@ static struct snd_soc_jack headset_jack; +enum PINCTRL_PIN_STATE { + PIN_STATE_DEFAULT = 0, + PIN_TDM_OUT_ON, + PIN_TDM_OUT_OFF, + PIN_STATE_MAX +}; + +static const char * const mt8183_pin_str[PIN_STATE_MAX] = { + "default", "aud_tdm_out_on", "aud_tdm_out_off", +}; + +struct mt8183_da7219_max98357_priv { + struct pinctrl *pinctrl; + struct pinctrl_state *pin_states[PIN_STATE_MAX]; +}; + static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -204,6 +220,47 @@ SND_SOC_DAILINK_DEFS(tdm, DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON])) + return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]); + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + + return ret; +} + +static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); +} + +static struct snd_soc_ops mt8183_da7219_tdm_ops = { + .startup = mt8183_da7219_tdm_startup, + .shutdown = mt8183_da7219_tdm_shutdown, +}; + static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { /* FE */ { @@ -353,6 +410,8 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_tdm_ops, SND_SOC_DAILINK_REG(tdm), }, }; @@ -409,7 +468,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_card *card = &mt8183_da7219_max98357_card; struct device_node *platform_node; struct snd_soc_dai_link *dai_link; - struct pinctrl *default_pins; + struct mt8183_da7219_max98357_priv *priv; int ret, i; card->dev = &pdev->dev; @@ -443,12 +502,45 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return ret; } - default_pins = - devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); - if (IS_ERR(default_pins)) { - dev_err(&pdev->dev, "%s set pins failed\n", + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + priv->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(priv->pinctrl)) { + dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", __func__); - return PTR_ERR(default_pins); + return PTR_ERR(priv->pinctrl); + } + + for (i = 0; i < PIN_STATE_MAX; i++) { + priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, + mt8183_pin_str[i]); + if (IS_ERR(priv->pin_states[i])) { + ret = PTR_ERR(priv->pin_states[i]); + dev_info(&pdev->dev, "%s Can't find pin state %s %d\n", + __func__, mt8183_pin_str[i], ret); + } + } + + if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); + } + + if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_DEFAULT]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); } return ret; -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 098/121] ASoC: Intel: Skylake: Enable codec wakeup during chip init
by Sasha Levin 12 Apr '20

12 Apr '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e603f11d5df8997d104ab405ff27640b90baffaa ] Follow the recommendation set by hda_intel.c and enable HDMI/DP codec wakeup during bus initialization procedure. Disable wakeup once init completes. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 58ba3e9469ba0..5cb6421de6fb2 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -130,6 +130,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) struct hdac_ext_link *hlink; int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); @@ -138,6 +139,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 097/121] ASoC: stm32: spdifrx: fix regmap status check
by Sasha Levin 12 Apr '20

12 Apr '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit a168dae5ea14283e8992d5282237bb0d6a3e1c06 ] Release resources when exiting on error. Fixes: 1a5c0b28fc56 ("ASoC: stm32: spdifrx: manage identification registers") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Link: https://lore.kernel.org/r/20200318144125.9163-2-olivier.moysan@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_spdifrx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 3769d9ce5dbef..e6e75897cce83 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -1009,6 +1009,8 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) if (idr == SPDIFRX_IPIDR_NUMBER) { ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver); + if (ret) + goto error; dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n", FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver), -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 066/121] ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed
by Sasha Levin 12 Apr '20

12 Apr '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit 8726ee6148fe24e2b29d4a961ad95c4ff8025d1d ] 1. Switch TDM GPIO pins according to playback on or off. 2. Pull TDM GPIO pins down when probed to avoid current leakage. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.2.I1d568b0c99742c6e755d051aadfd52e… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 104 +++++++++++++++++- 1 file changed, 98 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 43f99e59a0786..897a7bc4d3f3f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,6 +18,22 @@ static struct snd_soc_jack headset_jack; +enum PINCTRL_PIN_STATE { + PIN_STATE_DEFAULT = 0, + PIN_TDM_OUT_ON, + PIN_TDM_OUT_OFF, + PIN_STATE_MAX +}; + +static const char * const mt8183_pin_str[PIN_STATE_MAX] = { + "default", "aud_tdm_out_on", "aud_tdm_out_off", +}; + +struct mt8183_da7219_max98357_priv { + struct pinctrl *pinctrl; + struct pinctrl_state *pin_states[PIN_STATE_MAX]; +}; + static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -204,6 +220,47 @@ SND_SOC_DAILINK_DEFS(tdm, DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON])) + return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]); + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + + return ret; +} + +static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); +} + +static struct snd_soc_ops mt8183_da7219_tdm_ops = { + .startup = mt8183_da7219_tdm_startup, + .shutdown = mt8183_da7219_tdm_shutdown, +}; + static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { /* FE */ { @@ -353,6 +410,8 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_tdm_ops, SND_SOC_DAILINK_REG(tdm), }, }; @@ -409,7 +468,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_card *card = &mt8183_da7219_max98357_card; struct device_node *platform_node; struct snd_soc_dai_link *dai_link; - struct pinctrl *default_pins; + struct mt8183_da7219_max98357_priv *priv; int ret, i; card->dev = &pdev->dev; @@ -443,12 +502,45 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return ret; } - default_pins = - devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); - if (IS_ERR(default_pins)) { - dev_err(&pdev->dev, "%s set pins failed\n", + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + priv->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(priv->pinctrl)) { + dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", __func__); - return PTR_ERR(default_pins); + return PTR_ERR(priv->pinctrl); + } + + for (i = 0; i < PIN_STATE_MAX; i++) { + priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, + mt8183_pin_str[i]); + if (IS_ERR(priv->pin_states[i])) { + ret = PTR_ERR(priv->pin_states[i]); + dev_info(&pdev->dev, "%s Can't find pin state %s %d\n", + __func__, mt8183_pin_str[i], ret); + } + } + + if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); + } + + if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_DEFAULT]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); } return ret; -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 120/149] ASoC: Intel: Skylake: Enable codec wakeup during chip init
by Sasha Levin 12 Apr '20

12 Apr '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e603f11d5df8997d104ab405ff27640b90baffaa ] Follow the recommendation set by hda_intel.c and enable HDMI/DP codec wakeup during bus initialization procedure. Disable wakeup once init completes. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f755ca2484cff..0f3f291c223e2 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -130,6 +130,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) struct hdac_ext_link *hlink; int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); @@ -138,6 +139,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 119/149] ASoC: stm32: spdifrx: fix regmap status check
by Sasha Levin 12 Apr '20

12 Apr '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit a168dae5ea14283e8992d5282237bb0d6a3e1c06 ] Release resources when exiting on error. Fixes: 1a5c0b28fc56 ("ASoC: stm32: spdifrx: manage identification registers") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Link: https://lore.kernel.org/r/20200318144125.9163-2-olivier.moysan@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_spdifrx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 3769d9ce5dbef..e6e75897cce83 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -1009,6 +1009,8 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) if (idr == SPDIFRX_IPIDR_NUMBER) { ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver); + if (ret) + goto error; dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n", FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver), -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 086/149] ASoC: soc-pcm: fix regression in soc_new_pcm()
by Sasha Levin 12 Apr '20

12 Apr '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit a4877a6fb2bd2e356a5eaacd86d6b6d69ff84e69 ] Commit af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") swapped the SNDRV_PCM_STREAM_* parameter in the snd_soc_dai_stream_valid(cpu_dai, ...) checks. But that works only for codec2codec links. For normal links it breaks registration of playback/capture-only PCM devices. E.g. on qcom/apq8016_sbc there is usually one playback-only and one capture-only PCM device, but they disappeared after the commit. The codec2codec case was added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well") as an extra check (e.g. `playback = playback && cpu_playback->channels_min`). We should be able to simplify the code by checking directly for the correct stream type in the loop. This also fixes the regression because we check for PLAYBACK for both codec and cpu dai again when codec2codec is not used. Fixes: af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Tested-by: Jerome Brunet <jbrunet(a)baylibre.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Cc: Jerome Brunet <jbrunet(a)baylibre.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20200218103824.26708-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca01..a08ebbf5a37f3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2888,22 +2888,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { /* Adapt stream for codec2codec links */ - struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ? - &cpu_dai->driver->playback : &cpu_dai->driver->capture; - struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ? - &cpu_dai->driver->capture : &cpu_dai->driver->playback; + int cpu_capture = rtd->dai_link->params ? + SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int cpu_playback = rtd->dai_link->params ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, cpu_capture)) capture = 1; } - - capture = capture && cpu_capture->channels_min; - playback = playback && cpu_playback->channels_min; } if (rtd->dai_link->playback_only) { -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 081/149] ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed
by Sasha Levin 12 Apr '20

12 Apr '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit 8726ee6148fe24e2b29d4a961ad95c4ff8025d1d ] 1. Switch TDM GPIO pins according to playback on or off. 2. Pull TDM GPIO pins down when probed to avoid current leakage. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.2.I1d568b0c99742c6e755d051aadfd52e… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 104 +++++++++++++++++- 1 file changed, 98 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c65493721e90d..a3974b0a90157 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,6 +18,22 @@ static struct snd_soc_jack headset_jack; +enum PINCTRL_PIN_STATE { + PIN_STATE_DEFAULT = 0, + PIN_TDM_OUT_ON, + PIN_TDM_OUT_OFF, + PIN_STATE_MAX +}; + +static const char * const mt8183_pin_str[PIN_STATE_MAX] = { + "default", "aud_tdm_out_on", "aud_tdm_out_off", +}; + +struct mt8183_da7219_max98357_priv { + struct pinctrl *pinctrl; + struct pinctrl_state *pin_states[PIN_STATE_MAX]; +}; + static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -204,6 +220,47 @@ SND_SOC_DAILINK_DEFS(tdm, DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON])) + return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]); + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + + return ret; +} + +static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); +} + +static struct snd_soc_ops mt8183_da7219_tdm_ops = { + .startup = mt8183_da7219_tdm_startup, + .shutdown = mt8183_da7219_tdm_shutdown, +}; + static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { /* FE */ { @@ -353,6 +410,8 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_tdm_ops, SND_SOC_DAILINK_REG(tdm), }, }; @@ -409,7 +468,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_card *card = &mt8183_da7219_max98357_card; struct device_node *platform_node; struct snd_soc_dai_link *dai_link; - struct pinctrl *default_pins; + struct mt8183_da7219_max98357_priv *priv; int ret, i; card->dev = &pdev->dev; @@ -443,12 +502,45 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return ret; } - default_pins = - devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); - if (IS_ERR(default_pins)) { - dev_err(&pdev->dev, "%s set pins failed\n", + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + priv->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(priv->pinctrl)) { + dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", __func__); - return PTR_ERR(default_pins); + return PTR_ERR(priv->pinctrl); + } + + for (i = 0; i < PIN_STATE_MAX; i++) { + priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, + mt8183_pin_str[i]); + if (IS_ERR(priv->pin_states[i])) { + ret = PTR_ERR(priv->pin_states[i]); + dev_info(&pdev->dev, "%s Can't find pin state %s %d\n", + __func__, mt8183_pin_str[i], ret); + } + } + + if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); + } + + if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_DEFAULT]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); } return ret; -- 2.20.1

1 0

[PATCH] ALSA: hda: add autodetection for SoundWire
by Pierre-Louis Bossart 11 Apr '20

11 Apr '20

When an ACPI companion device is present and the SoundWire link mask information is available, use SoundWire instead of legacy HDA or Skylake drivers. The SOF driver is selected when SoundWire or DMIC are detected. There is no precedence at this level. In the SOF driver proper, SoundWire will be handled first since it is mutually exclusive with HDaudio. Known devices with an existing DMI quirk bypass this detection to avoid any dependency on ACPI/DSDT tables. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 97 ++++++++++++++++++++++++++++++------ 1 file changed, 81 insertions(+), 16 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index be1df80ed013..1c1fdab2a55f 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -1,10 +1,13 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> +#include <linux/acpi.h> #include <linux/bits.h> #include <linux/dmi.h> #include <linux/module.h> #include <linux/pci.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/core.h> #include <sound/intel-dsp-config.h> #include <sound/intel-nhlt.h> @@ -14,9 +17,13 @@ static int dsp_driver; module_param(dsp_driver, int, 0444); MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); -#define FLAG_SST BIT(0) -#define FLAG_SOF BIT(1) -#define FLAG_SOF_ONLY_IF_DMIC BIT(16) +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) +#define FLAG_SOF_ONLY_IF_SOUNDWIRE BIT(17) + +#define FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE (FLAG_SOF_ONLY_IF_DMIC | \ + FLAG_SOF_ONLY_IF_SOUNDWIRE) struct config_entry { u32 flags; @@ -166,7 +173,7 @@ static const struct config_entry config_table[] = { } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x9dc8, }, #endif @@ -187,7 +194,7 @@ static const struct config_entry config_table[] = { } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa348, }, #endif @@ -204,18 +211,50 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "09C6") + }, + }, + { + /* early version of SKU 09C6 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0983") + }, + }, {} } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x02c8, }, #endif /* Cometlake-H */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF, + .device = 0x06c8, + .dmi_table = (const struct dmi_system_id []) { + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"), + }, + }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"), + }, + }, + {} + } + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x06c8, }, #endif @@ -236,7 +275,7 @@ static const struct config_entry config_table[] = { } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x34c8, }, #endif @@ -256,9 +295,8 @@ static const struct config_entry config_table[] = { {} } }, - { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa0c8, }, #endif @@ -303,6 +341,28 @@ static int snd_intel_dsp_check_dmic(struct pci_dev *pci) return ret; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +static int snd_intel_dsp_check_soundwire(struct pci_dev *pci) +{ + struct sdw_intel_acpi_info info; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(&pci->dev); + + ret = sdw_intel_acpi_scan(handle, &info); + if (ret < 0) + return ret; + + return info.link_mask; +} +#else +static int snd_intel_dsp_check_soundwire(struct pci_dev *pci) +{ + return 0; +} +#endif + int snd_intel_dsp_driver_probe(struct pci_dev *pci) { const struct config_entry *cfg; @@ -336,14 +396,18 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) return SND_INTEL_DSP_DRIVER_ANY; if (cfg->flags & FLAG_SOF) { - if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { - if (snd_intel_dsp_check_dmic(pci)) { - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); - return SND_INTEL_DSP_DRIVER_SOF; - } - } else { + if (cfg->flags & FLAG_SOF_ONLY_IF_SOUNDWIRE && + snd_intel_dsp_check_soundwire(pci) > 0) { + dev_info(&pci->dev, "SoundWire enabled on CannonLake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC && + snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); return SND_INTEL_DSP_DRIVER_SOF; } + if (!(cfg->flags & FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE)) + return SND_INTEL_DSP_DRIVER_SOF; } if (cfg->flags & FLAG_SST) @@ -355,3 +419,4 @@ EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Intel DSP config driver"); +MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT); -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek - Enable the headset mic on Asus FX505DT
by Adam Barber 11 Apr '20

11 Apr '20

On Asus FX505DT with Realtek ALC233, the headset mic is connected to pin 0x19, with default 0x411111f0. Enable headset mic by reconfiguring the pin to an external mic associated with the headphone on 0x21. Mic jack detection was also found to be working. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207131 Signed-off-by: Adam Barber <barberadam995(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f66a48154a57..5016fa50956a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7252,6 +7252,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), -- 2.26.0

2 1

Re: [PATCH] ALSA: hda/realtek - Enable the headset mic on Asus FX505DT
by Takashi Iwai 10 Apr '20

10 Apr '20

On Fri, 10 Apr 2020 02:55:33 +0200, Adam Barber wrote: > > On Asus FX505DT with Realtek ALC233, the headset mic is connected > to pin 0x19, with default 0x411111f0. > > Enable headset mic by reconfiguring the pin to an external mic > associated with the headphone on 0x21. Mic jack detection was also > found to be working. Could you give your sign-off? It's mandatory for merging to upstream. Also, if there is some information linke (e.g. Bugzilla), put it in "BugLink: $URL" line, too. thanks, Takashi > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index f66a48154a57..5016fa50956a 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -7252,6 +7252,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), > SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), > SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), > + SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), > SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), > SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), > SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), > -- > 2.26.0 >

1 0

[PATCH] ASoC: soc-core: Remove name length truncation
by Tom Eccles 09 Apr '20

09 Apr '20

snd_soc_dai.name and snd_soc_component.name are a simplified version of the parent device's name, truncated to NAME_SIZE characters (see fmt_single_name). NAME_SIZE was 32 characters. Some device names may share a common prefix longer than 32 characters, which leads to non-unique snd_soc_dai.name and snd_soc_component.name. When these are added to debugfs, the .name is used for the file. If the names are not unique, adding to debugfs fails. Instead, dynamically allocate name buffers to be as long as needed. Signed-off-by: Tom Eccles <tom.eccles(a)codethink.co.uk> --- sound/soc/soc-core.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 843b8b1c89d4..7803222ede54 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -43,8 +43,6 @@ #define CREATE_TRACE_POINTS #include <trace/events/asoc.h> -#define NAME_SIZE 32 - static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); static LIST_HEAD(unbind_card_list); @@ -2291,13 +2289,15 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); */ static char *fmt_single_name(struct device *dev, int *id) { - char *found, name[NAME_SIZE]; + char *found, *name; int id1, id2; if (dev_name(dev) == NULL) return NULL; - strlcpy(name, dev_name(dev), NAME_SIZE); + name = devm_kstrdup(dev, dev_name(dev), GFP_KERNEL); + if (!name) + return NULL; /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); @@ -2313,20 +2313,23 @@ static char *fmt_single_name(struct device *dev, int *id) } else { /* I2C component devices are named "bus-addr" */ if (sscanf(name, "%x-%x", &id1, &id2) == 2) { - char tmp[NAME_SIZE]; + char *tmp; /* create unique ID number from I2C addr and bus */ *id = ((id1 & 0xffff) << 16) + id2; /* sanitize component name for DAI link creation */ - snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, - name); - strlcpy(name, tmp, NAME_SIZE); + tmp = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", + dev->driver->name, name); + devm_kfree(dev, name); + if (!tmp) + return NULL; + name = tmp; } else *id = 0; } - return devm_kstrdup(dev, name, GFP_KERNEL); + return name; } /* base-commit: fff876253c1e1b329257ba33045195c93e25adcd -- 2.20.1

1 0

Re: [PATCH] ALSA: ctxfi: Remove unnecessary cast in kfree
by Takashi Iwai 09 Apr '20

09 Apr '20

On Thu, 09 Apr 2020 13:20:52 +0200, Xu Wang wrote: > > Remove unnecassary casts in the argument to kfree. > > Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> Applied, thanks. Takashi

1 0

[PATCH v2 0/6] ASoC: topology: Propagate error appropriately
by Amadeusz Sławiński 09 Apr '20

09 Apr '20

v1: Check if kstrdup succeeded. v2: Remove unneeded freeing, which is performed in another place by dobj handlers. Additionally for functions which have return status which was ignored, perform success checks and handle failures in appropriate way. Amadeusz Sławiński (6): ASoC: topology: Add missing memory checks ASoC: topology: Check return value of soc_tplg_create_tlv ASoC: topology: Check return value of soc_tplg_*_create ASoC: topology: Check soc_tplg_add_route return value ASoC: topology: Check return value of pcm_new_ver ASoC: topology: Check return value of soc_tplg_dai_config sound/soc/soc-topology.c | 113 ++++++++++++++++++++++++++++++--------- 1 file changed, 88 insertions(+), 25 deletions(-) -- 2.17.1

5 23

alsa-gobject: Add shared library to operate ALSA rawmidi character device
by GitHub pull_request - opened 09 Apr '20

09 Apr '20

alsa-project/alsa-gobject pull request #16 was opened from takaswie: This patchset is to add a new library, libalsarawmidi, to operate ALSA rawmidi character device. This library is compatible with GObject Introspection. ALSA rawmidi interface is designed to transfer MIDI messages. Userspace application operate ALSA rawmidi character device for incoming/outgoing transmission. The transmission line is abstracted as substream. One character device is associated to several substreams. The application can indicate subdevice to select a substream or a pair of substreams from them. The library has ALSARawmidi.StreamPair to operate the character device. Userspace application call ALSARawmidi.StreamPair.open() to open the chracter device, select subdevice and attach the substreams to the device. After the call, data transmission already starts. The application can call ALSARawmidi.StreamPair.write_to_substream() to transfer MIDI message bytes. The application can also call ALSARawmidi.StreamPair.read_from_substream() to receive MIDI message bytes, but this call can be blocked till any messages are actually received and available as a default. For polling, GSource which ALSARawmidi.StreamPair.create_source() returns is useful. The application can use it for GMainContext/GMainLoop. When any message is available, ALSARawmidi.StreamPair emits 'handle-messages' GObject signal. The application register its handler to the signal in advance, then process the messages. ALSARawmidi.StreamPair has the other methods to drain/drop running substream: - ALSARawmidi.StreamPair.drain_substream() - ALSARawmidi.StreamPair.drop_substream() ALSARawmidi.StreamPair has more methods to return GObject-derived objects for information/status/parameters of substream: - ALSARawmidi.StreamPair.get_substream_info() - ALSARawmidi.StreamPair.get_substream_status() - ALSARawmidi.StreamPair.set_substream_params() The library includes global method to query information about rawmidi: - ALSARawmidi.get_rawmidi_sysname() - ALSARawmidi.get_rawmidi_devnode() - ALSARawmidi.get_device_id_list() - ALSARawmidi.get_subdevice_id_list() - ALSARawmidi.get_substream_info() Request URL : https://github.com/alsa-project/alsa-gobject/pull/16 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/16.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 1/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 09 Apr '20

09 Apr '20

snd_soc_runtime_activate() and snd_soc_runtime_deactivate() require locked pcm_mutex. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 392a1c5b15d3..42d416ac7e9b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -207,7 +207,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -285,7 +287,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) else stream = SNDRV_PCM_STREAM_CAPTURE; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; -- 2.21.0

3 5

[PATCH] ASoC: rt5682: fix building without I2C
by Arnd Bergmann 08 Apr '20

08 Apr '20

When I2C is disabled, building rt5682 produces a compile-time error: sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class 3716 | module_i2c_driver(rt5682_i2c_driver); | ^~~~~~~~~~~~~~~~~ sound/soc/codecs/rt5682.c:3716:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration sound/soc/codecs/rt5682.c:3706:26: error: 'rt5682_i2c_driver' defined but not used [-Werror=unused-variable] 3706 | static struct i2c_driver rt5682_i2c_driver = { | ^~~~~~~~~~~~~~~~~ Move the i2c specific entry points into an #ifdef section so it can actually be built standalone. Fixes: 5549ea647997 ("ASoC: rt5682: fix unmet dependencies") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/rt5682.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index c9268a230daa..a6548f870807 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -3224,12 +3224,6 @@ static const struct regmap_config rt5682_regmap = { .use_single_write = true, }; -static const struct i2c_device_id rt5682_i2c_id[] = { - {"rt5682", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); - static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) { @@ -3526,7 +3520,7 @@ int rt5682_io_init(struct device *dev, struct sdw_slave *slave) EXPORT_SYMBOL_GPL(rt5682_io_init); #endif -static int rt5682_i2c_probe(struct i2c_client *i2c, +static int __maybe_unused rt5682_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev); @@ -3680,13 +3674,14 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } -static void rt5682_i2c_shutdown(struct i2c_client *client) +static void __maybe_unused rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); rt5682_reset(rt5682); } +#ifdef CONFIG_I2C #ifdef CONFIG_OF static const struct of_device_id rt5682_of_match[] = { {.compatible = "realtek,rt5682i"}, @@ -3703,6 +3698,12 @@ static const struct acpi_device_id rt5682_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match); #endif +static const struct i2c_device_id rt5682_i2c_id[] = { + {"rt5682", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); + static struct i2c_driver rt5682_i2c_driver = { .driver = { .name = "rt5682", @@ -3714,6 +3715,7 @@ static struct i2c_driver rt5682_i2c_driver = { .id_table = rt5682_i2c_id, }; module_i2c_driver(rt5682_i2c_driver); +#endif MODULE_DESCRIPTION("ASoC RT5682 driver"); MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); -- 2.26.0

2 1

[PATCH] ASoC: brcm: fix dma_addr_t cast to pointer
by Arnd Bergmann 08 Apr '20

08 Apr '20

A dma_addr_t is not a pointer and a cast causes a warning when the sizes are different: sound/soc/bcm/bcm63xx-pcm-whistler.c: In function 'bcm63xx_pcm_pointer': sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast] 184 | if ((void *)prtd->dma_addr_next == NULL) Change the comparison to NULL to an equivalent if() check that does not warn. Fixes: 88eb404ccc3e ("ASoC: brcm: Add DSL/PON SoC audio driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/bcm/bcm63xx-pcm-whistler.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index e46c390683e7..b7a1efc7406e 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -181,7 +181,7 @@ bcm63xx_pcm_pointer(struct snd_soc_component *component, snd_pcm_uframes_t x; struct bcm63xx_runtime_data *prtd = substream->runtime->private_data; - if ((void *)prtd->dma_addr_next == NULL) + if (!prtd->dma_addr_next) prtd->dma_addr_next = substream->runtime->dma_addr; x = bytes_to_frames(substream->runtime, -- 2.26.0

1 0

[GIT PULL] ASoC fixes for v5.7
by Mark Brown 08 Apr '20

08 Apr '20

The following changes since commit 3d2cdb854659851d991f5b8e97e847e3fd240625: Merge series "ASoC: Intel: add SoundWire machine driver" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: (2020-03-27 17:28:36 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.7 for you to fetch changes up to ccfc531695f3a4aada042f6bdb33ac6be24e1aec: ASoC: cs4270: pull reset GPIO low then high (2020-04-07 15:29:54 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.7 A collection of fixes that have been accumilated since the merge window, mainly relating to x86 platform support. ---------------------------------------------------------------- Akshu Agrawal (1): ASoC: AMD: Clear format bits before setting them Cezary Rojewski (4): ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link Fabio Estevam (1): dt-bindings: sound: cs42l51: Remove unneeded I2C unit name Hans de Goede (6): ASoC: SOF: Turn "firmware boot complete" message into a dbg message ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map() ASoC: Intel: atom: Check drv->lock is locked in sst_fill_and_send_cmd_unlocked ASoC: Intel: atom: Fix uninitialized variable compiler warning ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet ASoC: rt5645: Add platform-data for Medion E1239T Julia Lawall (1): ASoC: stm32: sai: Add missing cleanup Mark Brown (1): Merge series "ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai link" from Cezary Rojewski <cezary.rojewski(a)intel.com>: Mike Willard (1): ASoC: cs4270: pull reset GPIO low then high Pierre-Louis Bossart (1): ASoC: soc-dai: fix DAI startup/shutdown sequence Stephan Gerhold (1): ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag Takashi Iwai (1): ASoC: bcm: Fix pointer cast warning YueHaibing (1): ASoC: rt5682: Fix build error without CONFIG_I2C 이경택 (4): ASoC: fix regwmask ASoC: dapm: connect virtual mux with default value ASoC: dpcm: allow start or stop during pause for backend ASoC: topology: use name_prefix for new kcontrol .../devicetree/bindings/sound/cirrus,cs42l51.yaml | 2 +- include/sound/soc-dai.h | 2 +- sound/soc/amd/raven/acp3x-i2s.c | 1 + sound/soc/amd/raven/acp3x.h | 2 ++ sound/soc/bcm/bcm63xx-pcm-whistler.c | 2 +- sound/soc/codecs/cs4270.c | 40 +++++++++++++++++++--- sound/soc/codecs/rt5645.c | 8 +++++ sound/soc/codecs/rt5682.c | 5 ++- sound/soc/intel/atom/sst-atom-controls.c | 4 +++ sound/soc/intel/atom/sst/sst_pvt.c | 2 +- sound/soc/intel/boards/bdw-rt5650.c | 1 - sound/soc/intel/boards/bdw-rt5677.c | 1 - sound/soc/intel/boards/broadwell.c | 1 - sound/soc/intel/boards/bytcr_rt5640.c | 11 ++++++ sound/soc/intel/boards/haswell.c | 1 - sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +-- sound/soc/soc-dai.c | 8 ++--- sound/soc/soc-dapm.c | 8 ++++- sound/soc/soc-ops.c | 4 +-- sound/soc/soc-pcm.c | 6 ++-- sound/soc/soc-topology.c | 2 +- sound/soc/sof/loader.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 4 ++- 23 files changed, 93 insertions(+), 28 deletions(-)

2 1

[PATCH 1/2] ALSA: usb-audio: Add mixer workaround for TRX40 and co
by Takashi Iwai 08 Apr '20

08 Apr '20

Some recent boards (supposedly with a new AMD platform) contain the USB audio class 2 device that is often tied with HD-audio. The device exposes an Input Gain Pad control (id=19, control=12) but this node doesn't behave correctly, returning an error for each inquiry of GET_MIN and GET_MAX that should have been mandatory. As a workaround, simply ignore this node by adding a usbmix_name_map table entry. The currently known devices are: * 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi * 0b05:1916 - ASUS ROG Zenith II * 0b05:1917 - ASUS ROG Strix * 0db0:0d64 - MSI TRX40 Creator * 0db0:543d - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 5ebca8013840..72b575c34860 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -359,6 +359,14 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { { 0 } }; +/* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX + * response for Input Gain Pad (id=19, control=12). Skip it. + */ +static const struct usbmix_name_map asus_rog_map[] = { + { 19, NULL, 12 }, /* FU, Input Gain Pad */ + {} +}; + /* * Control map entries */ @@ -488,6 +496,26 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1b1c, 0x0a42), .map = corsair_virtuoso_map, }, + { /* Gigabyte TRX40 Aorus Pro WiFi */ + .id = USB_ID(0x0414, 0xa002), + .map = asus_rog_map, + }, + { /* ASUS ROG Zenith II */ + .id = USB_ID(0x0b05, 0x1916), + .map = asus_rog_map, + }, + { /* ASUS ROG Strix */ + .id = USB_ID(0x0b05, 0x1917), + .map = asus_rog_map, + }, + { /* MSI TRX40 Creator */ + .id = USB_ID(0x0db0, 0x0d64), + .map = asus_rog_map, + }, + { /* MSI TRX40 */ + .id = USB_ID(0x0db0, 0x543d), + .map = asus_rog_map, + }, { 0 } /* terminator */ }; -- 2.25.0

1 1

[PATCH] ALSA: hda/realtek - Add quirk for MSI GL63
by Takashi Iwai 08 Apr '20

08 Apr '20

MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 019239190f6d..de2826f90d34 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2455,6 +2455,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.25.0

1 0

Error compiling alsatplg
by GitHub issues - opened 08 Apr '20

08 Apr '20

alsa-project/alsa-utils issue #37 was opened from dbaluta: With latest alsa-lib and alsa-utils sources I cannot compile alsatplg utility. Steps followed: # Compile alsa-lib 1. ./gitcompile --prefix=$HOME/work/sof_dir/alsa-lib-bins/ 2. make install #custom instalation to be later used by alsa-utils. # Compile alsa-utils 1. /gitcompile --with-alsa-inc-prefix=$HOME/work/sof_dir/alsa-lib-bins/include/ --with-alsa-prefix=$HOME/work/sof_dir/alsa-lib-bins/lib 2. I get the following errors: ``` -ldl -lpthread topology.o: In function `load_topology': /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:135: undefined reference to `snd_tplg_create' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:141: undefined reference to `snd_tplg_load' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:145: undefined reference to `snd_tplg_free' topology.o: In function `version': /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:63: undefined reference to `snd_tplg_version' topology.o: In function `dump': /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:218: undefined reference to `snd_tplg_save' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:219: undefined reference to `snd_tplg_free' topology.o: In function `compile': /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:245: undefined reference to `snd_tplg_build_bin' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:246: undefined reference to `snd_tplg_free' topology.o: In function `decode': /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:268: undefined reference to `snd_tplg_create' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:273: undefined reference to `snd_tplg_decode' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:281: undefined reference to `snd_tplg_save' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:282: undefined reference to `snd_tplg_free' /home/nxf25322/work/sof_dir/alsa-utils/topology/topology.c:276: undefined reference to `snd_tplg_free' collect2: error: ld returned 1 exit status Makefile:433: recipe for target 'alsatplg' failed ``` Similar with https://github.com/alsa-project/alsa-utils/issues/33 Issue URL : https://github.com/alsa-project/alsa-utils/issues/37 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

M-Audio Fast Track Ultra S/PDIF clock source
by George Marques 07 Apr '20

07 Apr '20

Hello, I have my trusty old FTU audio interface and I've been super happy of how it works well on Linux with ALSA. Today I finally found some use for the S/PDIF input, which would allow me to free up two analog inputs in my setup. However, I found out it's not possible to change the clock source using ALSA, so I cannot use the S/PDIF input on Linux. This issue was raised by Felix Homann back in 2011: https://www.alsa-project.org/pipermail/alsa-devel/2011-June/040846.html but I couldn't find any new development, even though other issues were fixed. On Windows the proprietary driver naturally has an option for that, which allowed me to verify that the input is in fact working correctly. Changing on Windows and then rebooting into Linux does not work, so either this is not persisted or something on Linux is unknowingly changing it back. I actually want to help add this option to the driver, so I can use it myself but also make it available to others who still rely on this interface. My questions are: 1) Was there any progress in this regard? Most of work has been done around 2012 for what I could find, nothing newer, so can I assume no work was done after the concern was raised? 2) If I were to develop this option, where could I start? I can get myself around with C, but I have no idea how to debug and test my patches, nor how to find the exact control. I know it's an old card, but it works very well for me so I have no excuse to just replace it at this point. I hope someone can help me.

1 0

TRX40 Realtek ALC1220-VB USB Audio
by laurent Trinques 07 Apr '20

07 Apr '20

Hello, sudo dmesg -lerr -lcrit [ 0.386478] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT1A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386484] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386488] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT1A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386490] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386493] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT2A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386495] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386497] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT2A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386499] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386501] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT3A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386504] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386506] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT3A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386508] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386510] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT4A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386513] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386516] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT4A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386518] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386520] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT5A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386523] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386526] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT1B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386529] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386531] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT1B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386534] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386536] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT2B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386538] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386541] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT2B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386543] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386546] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT3B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386548] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386550] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT3B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386553] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386555] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT4B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386557] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386560] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT4B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386562] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386565] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT5B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386567] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386571] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT1C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386573] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386575] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT1C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386578] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386580] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT2C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386583] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386585] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT2C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386587] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386590] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT3C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386592] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386595] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT3C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386597] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386599] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT4C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386602] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386604] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT4C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386606] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386609] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT5C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386611] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386615] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT1D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386617] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386620] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT1D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386622] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386624] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT2D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386627] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386629] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT2D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386632] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386634] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT3D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386636] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386639] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT3D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386641] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386644] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT4D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386646] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386648] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT4D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386651] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386653] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT5D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386656] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 2.527434] sp5100-tco sp5100-tco: Watchdog hardware is disabled [ 2.545332] Error: Driver 'pcspkr' is already registered, aborting... [ 2.572542] snd_hda_intel 0000:23:00.4: no codecs found! [ 8.169209] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.169214] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.430796] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.430800] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.431164] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.483290] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.483294] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.483662] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.628066] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.628071] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.628557] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.849035] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.849039] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.849405] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.040530] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.040534] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.040903] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.233150] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.233154] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.233522] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.425023] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.425027] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.425389] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.617141] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.617145] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.617636] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.809016] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.809020] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.809385] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.000383] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.000387] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 10.000754] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.417141] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.417146] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 10.417641] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.467123] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.467127] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 10.467870] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.148540] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.148545] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.149030] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.211031] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.211035] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.211402] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.289284] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.289289] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.289654] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.480906] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.480911] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.481275] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.673023] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.673028] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.673392] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.864772] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.864777] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.865138] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.057144] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.057149] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.057639] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.248638] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.248643] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.249010] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.440883] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.440887] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.441251] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.633004] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.633009] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.633373] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 16.016995] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 16.017000] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 16.017491] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 Best regards, Laurent

2 4

[PATCH 0/2] ALSA: Fix for bad ctl access patterns
by Takashi Iwai 07 Apr '20

07 Apr '20

Hi, the recently introduced ctl access validator succeessfully caught some real issues. Here are some fixes. Takashi === Takashi Iwai (2): ALSA: hda: Fix potential access overflow in beep helper ALSA: ice1724: Fix invalid access for enumerated ctl items sound/pci/hda/hda_beep.c | 6 +++++- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 2 files changed, 7 insertions(+), 3 deletions(-) -- 2.25.0

2 3

[PATCH] ASoC: cs4270: pull reset GPIO low then high
by Mike Willard 07 Apr '20

07 Apr '20

Pull the RST line low then high when initializing the driver, in order to force a reset of the chip. Previously, the line was not pulled low, which could result in the chip registers not resetting to their default values on boot. Signed-off-by: Mike Willard <mwillard(a)izotope.com> --- sound/soc/codecs/cs4270.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5f25b9f872bd..8a02791e44ad 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -137,6 +137,9 @@ struct cs4270_private { /* power domain regulators */ struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; + + /* reset gpio */ + struct gpio_desc *reset_gpio; }; static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { @@ -648,6 +651,22 @@ static const struct regmap_config cs4270_regmap = { .volatile_reg = cs4270_reg_is_volatile, }; +/** + * cs4270_i2c_remove - deinitialize the I2C interface of the CS4270 + * @i2c_client: the I2C client object + * + * This function puts the chip into low power mode when the i2c device + * is removed. + */ +static int cs4270_i2c_remove(struct i2c_client *i2c_client) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); + + gpiod_set_value_cansleep(cs4270->reset_gpio, 0); + + return 0; +} + /** * cs4270_i2c_probe - initialize the I2C interface of the CS4270 * @i2c_client: the I2C client object @@ -660,7 +679,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs4270_private *cs4270; - struct gpio_desc *reset_gpiod; unsigned int val; int ret, i; @@ -679,10 +697,21 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, if (ret < 0) return ret; - reset_gpiod = devm_gpiod_get_optional(&i2c_client->dev, "reset", - GPIOD_OUT_HIGH); - if (PTR_ERR(reset_gpiod) == -EPROBE_DEFER) - return -EPROBE_DEFER; + /* reset the device */ + cs4270->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs4270->reset_gpio)) { + dev_dbg(&i2c_client->dev, "Error getting CS4270 reset GPIO\n"); + return PTR_ERR(cs4270->reset_gpio); + } + + if (cs4270->reset_gpio) { + dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); + gpiod_set_value_cansleep(cs4270->reset_gpio, 1); + } + + /* Sleep 500ns before i2c communications */ + ndelay(500); cs4270->regmap = devm_regmap_init_i2c(i2c_client, &cs4270_regmap); if (IS_ERR(cs4270->regmap)) @@ -735,6 +764,7 @@ static struct i2c_driver cs4270_i2c_driver = { }, .id_table = cs4270_id, .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, }; module_i2c_driver(cs4270_i2c_driver); -- 2.17.1

2 1

[PATCH v2 00/12] [RESEND] ASoC: SOF DSP virtualisation
by Guennadi Liakhovetski 07 Apr '20

07 Apr '20

This patch series extends the SOF driver to add support for DSP virtualisation to ASoC. It is built on top of VirtIO, contains a guest driver and a vhost in-kernel guest driver. This version supports a single playback and a single capture interface on the guest. The specific guest audio topology is supplied by the host from a file, specified on the QEMU command line. This solution is designed to use advanced DSP functionality, available on the host. In case no DSP is available on the host, a fall-back should be provided in the future to the pure audio virtualisation standard, currently developed by OASIS. Extensive documentation is available at [1]. I'd like to have at least a tentative approval on this series to send vhost patches 3 and 9 to virtualisation lists / maintainers for review and eventual merging. v2: one patch merged, one patch resubmitted separately, otherwise no changes. [1] https://thesofproject.github.io/latest/developer_guides/virtualization/virt… Guennadi Liakhovetski (12): ASoC: add function parameters to enable forced path pruning ASoC: SOF: extract firmware-related operation into a function ASoC: SOF: VirtIO: make a function global vhost: convert VHOST_VSOCK_SET_RUNNING to a generic ioctl ASoC: SOF: support IPC with immediate response ASoC: SOF: add a power status IPC ASoC: SOF: add two helper lookup functions ASoC: SOF: add a VirtIO DSP driver ASoC: SOF: add a vhost driver: sound part vhost: add an SOF DSP driver ASoC: SOF: VirtIO: free guest pipelines upon termination ASoC: SOF: VirtIO: enable simultaneous playback and capture drivers/vhost/Kconfig | 7 + drivers/vhost/Makefile | 5 + drivers/vhost/dsp.c | 731 ++++++++++++++++++++++ include/sound/soc-dpcm.h | 28 +- include/sound/soc-topology.h | 3 + include/sound/sof.h | 4 + include/sound/sof/header.h | 3 + include/sound/sof/topology.h | 1 + include/sound/sof/virtio.h | 206 +++++++ include/uapi/linux/vhost.h | 9 +- include/uapi/linux/vhost_types.h | 7 + include/uapi/linux/virtio_ids.h | 1 + sound/soc/soc-compress.c | 2 +- sound/soc/soc-dapm.c | 8 +- sound/soc/soc-pcm.c | 98 ++- sound/soc/sof/Kconfig | 8 + sound/soc/sof/Makefile | 8 + sound/soc/sof/core.c | 114 ++-- sound/soc/sof/ipc.c | 39 +- sound/soc/sof/loader.c | 4 + sound/soc/sof/ops.h | 10 +- sound/soc/sof/pcm.c | 13 +- sound/soc/sof/pm.c | 4 + sound/soc/sof/sof-audio.c | 33 + sound/soc/sof/sof-audio.h | 21 + sound/soc/sof/sof-priv.h | 52 ++ sound/soc/sof/topology.c | 71 ++- sound/soc/sof/vhost-be.c | 1248 ++++++++++++++++++++++++++++++++++++++ sound/soc/sof/virtio-fe.c | 922 ++++++++++++++++++++++++++++ 29 files changed, 3552 insertions(+), 108 deletions(-) create mode 100644 drivers/vhost/dsp.c create mode 100644 include/sound/sof/virtio.h create mode 100644 sound/soc/sof/vhost-be.c create mode 100644 sound/soc/sof/virtio-fe.c -- 1.9.3

3 21

alsa-gobject: Add shared library to operate ALSA hwdep character device
by GitHub pull_request - opened 07 Apr '20

07 Apr '20

alsa-project/alsa-gobject pull request #15 was opened from takaswie: This patchset is to add a new library, libalsahwdep, to operate ALSA hwdep character device. ALSA hwdep interface is designed to execute driver-dependent code from user space application. Therefore there're a few common operations. Furthermore, some of the operations are just for ALSA snd-usb-usx2y driver for TASCAM US-122, US-224 and US-428, as of Linux kernel v5.6. The library doesn't support the device-dependent operations. The library has 4 global APIS. ALSAHwdep.get_device_id_list() returns the list of numerical IDs of hwdep device in a card. ALSAHwdep.get_hwdep_sysname() and ALSAHwdep.get_hwdep_devnode() returns sysname and devnode for the card and hwdep device. ALSAHwdep.get_device_info() returns ALSAHwdep.DeviceInfo GObject-derived object which represents the information of hwdep device of the card. The object has some properties, one of which is a enumeration in ALSAHwdep.IfaceType to represent the type of hwdep device. Request URL : https://github.com/alsa-project/alsa-gobject/pull/15 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

New HP mute led feature
by Kailang 07 Apr '20

07 Apr '20

Hi Takashi, HP had new mute led feature. Attach two patches will enable it. BR, Kailang

2 4

[PATCH] ASoC: rt5645: Add platform-data for Medion E1239T
by Hans de Goede 06 Apr '20

06 Apr '20

The Medion E1239T uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like other models following Intel's Brasswell's reference design. This commit adds a DMI quirk pointing to the intel_braswell_platform_data for this model. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5645.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 92d67010aeed..6ba1849a77b0 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3758,6 +3758,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&kahlee_platform_data, }, + { + .ident = "Medion E1239T", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MEDION"), + DMI_MATCH(DMI_PRODUCT_NAME, "E1239T MD60568"), + }, + .driver_data = (void *)&intel_braswell_platform_data, + }, { } }; -- 2.26.0

3 2

[PATCH] ASoC: stm32: sai: Add missing cleanup
by Julia Lawall 06 Apr '20

06 Apr '20

The commit 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") converts some function calls to their non-devm equivalents. The appropriate cleanup code was added to the remove function, but not to the probe function. Add a call to snd_dmaengine_pcm_unregister to compensate for the call to snd_dmaengine_pcm_register in case of subsequent failure. Fixes: commit 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- Not tested. sound/soc/stm/stm32_sai_sub.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 2bd280c..0d0c9af 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1556,8 +1556,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) + if (ret) { + snd_dmaengine_pcm_unregister(&pdev->dev); return ret; + } if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif;

3 2

[PATCH v3] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Hans de Goede 06 Apr '20

06 Apr '20

The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6bd9ae813be2..d14d5f7db168 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.26.0

3 4

[PATCH] ASoC: qcom: common: Silence EPROBE_DEFER messages
by Bjorn Andersson 06 Apr '20

06 Apr '20

The Qualcomm sound card typically depends on several other drivers, resulting in plenty of noise in the kernel log due to probe deferral. Silence these errors. Signed-off-by: Bjorn Andersson <bjorn.andersson(a)linaro.org> --- sound/soc/qcom/common.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 6c20bdd850f3..cb266d7ec03e 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -80,7 +80,8 @@ int qcom_snd_parse_of(struct snd_soc_card *card) ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name); if (ret) { - dev_err(card->dev, "%s: error getting cpu dai name\n", link->name); + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "%s: error getting cpu dai name", link->name); goto err; } @@ -96,7 +97,8 @@ int qcom_snd_parse_of(struct snd_soc_card *card) ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); if (ret < 0) { - dev_err(card->dev, "%s: codec dai not found\n", link->name); + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "%s: codec dai not found\n", link->name); goto err; } link->no_pcm = 1; -- 2.24.0

2 1

alsa-gobject: integrate ALSASeq to handle queue operations
by GitHub pull_request - opened 06 Apr '20

06 Apr '20

alsa-project/alsa-gobject pull request #14 was opened from takaswie: In ALSA Sequencer, queue is an important element to dispatch scheduled events. Without queue, any events are going to be dispatched immediately and this mode is already supported. This patchset is to support queue operations and complete development for ALSASeq. The information of queue is represented by ALSASeq.QueueInfo GObject-derived object. Several APIs are added to ALSASeq.UserClient to maintain lifetime of queue: - ALSASeq.UserClient.create_queue() - ALSASeq.UserClient.delete_queue() - ALSASeq.UserClient.update_queue() The maintenance of actual queue behaviour can be already done by scheduling events with ALSASeq.EventDataQueue with appropriate event type. The status and tempo of queue to dispatch scheduled events is repsented by ALSASeq.QueueStatus and ALSASeq.QueueTempo. Some APIs are added to ALSASeq.UserClient to retrieve them: - ALSASeq.UserClient.set_queue_tempo() - ALSASeq.UserClient.get_queue_tempo() - ALSASeq.UserClient.set_queue_timer() - ALSASeq.UserClient.get_queue_timer() The registered queues can be reserved. Some APIs are added to ALSASeq.UserClient for the reservation: - ALSASeq.UserClient.get_queue_usage() - ALSASeq.UserClient.set_queue_usage() The queue is assigned to an instance of ALSA Timer device, practically. In this case, global system timer device is used according to parameters of snd_seq kernel module as a default. This global system timer is based on Linux timer wheel. The assignment is represented by subclasses of ALSASeq.QueueTimer and ALSASeq.QueueTimerALSA is currently available. Two APIs are added to ALSASeq.UserClient for the assignment: - ALSASeq.UserClient.set_queue_timer() - ALSASeq.UserClient.get_queue_timer() Finally, the scheduled events can be removed from the queues. The batch of target events is represented by ALSASeq.RemoveFilter and ALSASeq.UserClient.remove_events() is available for the removal. Request URL : https://github.com/alsa-project/alsa-gobject/pull/14 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

alsa-gobject: misc fixes
by GitHub pull_request - opened 06 Apr '20

06 Apr '20

alsa-project/alsa-gobject pull request #13 was opened from takaswie: This patchset is misc fixes. Request URL : https://github.com/alsa-project/alsa-gobject/pull/13 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/13.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/4] ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai link
by Cezary Rojewski 06 Apr '20

06 Apr '20

As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Link to first message in conversation: https://lkml.org/lkml/2020/3/18/54 Cezary Rojewski (4): ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link sound/soc/intel/boards/bdw-rt5650.c | 1 - sound/soc/intel/boards/bdw-rt5677.c | 1 - sound/soc/intel/boards/broadwell.c | 1 - sound/soc/intel/boards/haswell.c | 1 - 4 files changed, 4 deletions(-) -- 2.17.1

5 22

Re:
by Ruslan Bilovol 05 Apr '20

05 Apr '20

Hi, Please also add to CC related mailing lists (alsa-devel, linux-usb) rather then directly emailing - community may also help with the issue. Also it can be googled so if somebody else have same issue it can find answers faster. On Fri, Apr 3, 2020 at 10:56 AM Colin Williams <colin.williams.seattle(a)gmail.com> wrote: > > https://ubuntuforums.org/showthread.php?t=2439897 > > On Thu, Apr 2, 2020 at 4:50 PM Colin Williams <colin.williams.seattle(a)gmail.com> wrote: >> >> Hello, >> >> Is it possible that one of these commits or related broke support for the Blue Mic Yeti? >> >> https://github.com/torvalds/linux/blame/ac438771ccb4479528594c7e19f2c39cf18… Tha'ts workaround to ignore last altsetting which is the same as previous. During UAC3 implementation, I reimplemented that workaround carefully, but I didn't have (and still do not own) any Blue Mic USB device. I don't know whether it was tested after that by anyone. >> >> I am getting the following when I plug my mic in: Which kernel version is that? Have you tried latest Linux Kernel? >> >> [ 1283.848740] usb 1-1.2: new full-speed USB device number 82 using ehci-pci >> [ 1283.964802] usb 1-1.2: New USB device found, idVendor=b58e, idProduct=9e84, bcdDevice= 1.00 >> [ 1283.964808] usb 1-1.2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 >> [ 1283.964810] usb 1-1.2: Product: Yeti Stereo Microphone >> [ 1283.964812] usb 1-1.2: Manufacturer: Blue Microphones >> [ 1284.080671] usb 1-1.3: new low-speed USB device number 83 using ehci-pci >> [ 1284.784678] usb 1-1.3: device descriptor read/64, error -32 >> [ 1285.180674] usb 1-1.3: device descriptor read/64, error -32 >> [ 1285.992682] usb 1-1.3: new low-speed USB device number 84 using ehci-pci >> [ 1286.696672] usb 1-1.3: device descriptor read/64, error -32 >> [ 1287.092695] usb 1-1.3: device descriptor read/64, error -32 >> [ 1287.200804] usb 1-1-port3: attempt power cycle >> [ 1287.804662] usb 1-1.3: new low-speed USB device number 85 using ehci-pci >> [ 1288.220686] usb 1-1.3: device not accepting address 85, error -32 >> [ 1288.508685] usb 1-1.3: new low-speed USB device number 86 using ehci-pci >> [ 1288.924690] usb 1-1.3: device not accepting address 86, error -32 >> [ 1288.924916] usb 1-1-port3: unable to enumerate USB device >> [ 1288.925391] usb 1-1.2: USB disconnect, device number 82 >> [ 1289.308736] usb 1-1.3: new low-speed USB device number 87 using ehci-pci >> [ 1289.596727] usb 1-1.3: device descriptor read/64, error -32 >> [ 1289.992635] usb 1-1.3: device descriptor read/64, error -32 >> [ 1290.596683] usb 1-1.3: new low-speed USB device number 88 using ehci-pci >> [ 1290.888718] usb 1-1.3: device descriptor read/64, error -32 >> [ 1291.284673] usb 1-1.3: device descriptor read/64, error -32 >> [ 1291.392928] usb 1-1-port3: attempt power cycle Looking at this log, it seems the issue happens during enumeration, so mentioned workaround isn't executed yet at this moment. So it seems this is related to USB core, not to ALSA driver. Thanks, Ruslan >> >> Furthermore, there is some evidence this is happening to other users: >> >> https://askubuntu.com/questions/1027188/external-yeti-michrophone-not-detec… >> >> Best Regards, >> >> Colin Williams >> >>

2 1

[PATCH] ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Alpha S
by Emmanuel Pescosta 04 Apr '20

04 Apr '20

Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud Alpha S (0951:16d8) uses two interfaces, but only the second interface contains the capture stream. This patch delays the registration until the second interface appears. Signed-off-by: Emmanuel Pescosta <emmanuelpescosta099(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 86f192a3043d..a8ece1701068 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1827,6 +1827,7 @@ struct registration_quirk { static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ + REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ { 0 } /* terminator */ }; -- 2.26.0

2 1

alsa-gobject: fix annotations for constructors
by GitHub pull_request - opened 04 Apr '20

04 Apr '20

alsa-project/alsa-gobject pull request #12 was opened from takaswie: Some constructor functions have redundant or invalid annotations. Request URL : https://github.com/alsa-project/alsa-gobject/pull/12 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/12.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

alsa-gobject: normalize function signatures
by GitHub pull_request - opened 04 Apr '20

04 Apr '20

alsa-project/alsa-gobject pull request #11 was opened from takaswie: The alsa-gobject project has a loose convention to have functions returning nothing. This pull request refine according to it as well as fixes for some bugs. Request URL : https://github.com/alsa-project/alsa-gobject/pull/11 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/11.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by GitHub pull_request - opened 03 Apr '20

03 Apr '20

alsa-project/alsa-lib pull request #40 was opened from im-0: ASUS Xonar AE is a PCI-Express card containing USB controller: USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] and the actual USB sound card: ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD As other Xonar USB sound cards, it uses second device for digital output. Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/40 Patch URL : https://github.com/alsa-project/alsa-lib/pull/40.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/3] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Hans de Goede 03 Apr '20

03 Apr '20

sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f1..2c3798034b1d 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.26.0

3 6

alsa-gobject: add library to operate ALSA Sequencer character device without queue feature
by GitHub pull_request - edited 03 Apr '20

03 Apr '20

alsa-project/alsa-gobject pull request #10 was edited from takaswie: This request is to add a new library, libalsaseq, to operate ALSA Sequencer character device. In design of ALSA Sequencer, one character device is added to Linux system. Usually a special file, '/dev/snd/seq', is used for the character device. Userspace applications can operate ALSA Sequencer via the character device. The library supports functionalities of ALSA Sequencer but without queue-related ones. The library includes some GObject-derived object. ALSASeq.UserClient maintains file descriptor of ALSA Sequencer character device, thus it's the center of user operation. The other objects represent data structures in UAPI of ALSA Sequencer. * ALSASeqSystemInfo - struct snd_seq_system_info * ALSASeqClientInfo - struct snd_seq_client_info * ALSASeqClientPool - struct snd_seq_client_pool * ALSASeqPortInfo - struct snd_seq_port_info * ALSASeqSubscribeData - struct snd_seq_port_subscribe The library includes some GObject boxed object just to represent data structures. ALSASeq.Addr is a thin wrapper to struct snd_seq_addr, and ALSASeq.Tstamp is a thin wrapper to struct snd_seq_timestamp. For operations without instances of ALSASeq.UserClient, the library has some global APIs. ALSA Sequencer has some macros. They're represented as GObject enumerations and flags. * ALSASeq.ClientType - one of NO_CLIENT, USER_CLIENT, KERNEL_CLIENT * ALSASeq.FilterAttrFlag - SNDRV_SEQ_FILTER_XXX * ALSASeq.PortCapFlag - SNDRV_PORT_CAP_XXX * ALSASeq.PortAttrFlag - NDRV_SEQ_PORT_TYPE_XXX * ALSASeq.EventType - SNDRV_SEQ_EVENT_XXX * ALSASeq.EventTimestampMode - SNDRV_SEQ_TIME_STAMP_XXX * ALSASeq.EventTimeMode - SNDRV_SEQ_TIME_MODE_XXX * ALSASeq.EventLengthMode - SNDRV_SEQ_EVENT_LENGTH_XXX * ALSASeq.EventPriorityMode - SNDRV_SEQ_PRIORITY_XXX * ALSASeq.PortSubscribeFlag - SNDRV_SEQ_PORT_SUBS_XXX * ALSASeq.QuerySubscribeType - SNDRV_SEQ_QUERY_SUBS_XXX Some specific values for addressing are also represented by GObject enumerations and flags. * ALSASeq.SpecificAddress - SNDRV_SEQ_ADDRESS_XXX * ALSASeq.SpecificClientId - SNDRV_SEQ_CLIENT_XXX * ALSASeq.SpecificPortId - SNDRV_SEQ_PORT_SYSTEM_XXX * ALSASeq.SpecificQueueId - SNDRV_SEQ_QUEUE_XXX ALSA Sequencer has many types of event. The library has an GObject abstract object, ALSASeq.Event. Two objects are derived from it: * ALSASeq.EventFixed * ALSASeq.EventVariable The former has some ways to access to the data of event. The data is represented by GObject boxed object if it's complicated structure: * ALSASeq.EventDataConnect - struct snd_seq_connect * ALSASeq.EventDataNote - struct snd_seq_ev_note * ALSASeq.EventDataCtl - struct snd_seq_ev_ctrl * ALSASeq.EventDataResult - struct snd_seq_result * ALSASeq.EventDataQueue - struct snd_seq_ev_queue_control The latter is for events to have variable length of data blob. ALSASeq.UserClient has 'schedule_event' API to schedule arbitrary event. It also has 'handle-event' GObject signal for applications to handle asynchronous event as an instance of ALSASeq.EventFixed or ALSASeq.EventVariable. To handle the GObject signal, applications should get GSource from the instance and process GMainContext, typically by call of GMainLoop. The library doesn't have support for below structures since they're useless practically: * struct snd_seq_event_bounce * struct snd_seq_running_info In my later work, queue functions will be added. Request URL : https://github.com/alsa-project/alsa-gobject/pull/10 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/1] reviving ack callback for 5.7(latest)
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

ACK callback is useful to notify application pointer in DMA buffer to DSP, because it is called after application pointer is updated in ALSA. DSP can use the pointer to control preprocessing, detect underrun and so on. I'd like to revive it from 5.4 if it is possible. This patch is created in for-linus branch. Gyeongtaek Lee (1): ASoC: soc-component: add snd_soc_component_ack() include/sound/soc-component.h | 3 +++ sound/soc/soc-component.c | 14 ++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 19 insertions(+) base-commit: 52ac54e1de31af2d40f10dd4e711e774655e091b -- 2.21.0

3 2

[PATCH] quirk for Pioneer DJ DJM-250MK2 usb sound card
by František Kučera 03 Apr '20

03 Apr '20

Hello, I have created a quirk that adds support for the Pioneer DJ DJM-250MK2 mixer (acts like a USB sound card, but Vendor Specific Class). Output (from computer to card) is working: 8 channels, 48 000 Hz, S24_3LE. Input is not working. I tried, but I was unable to make it working even under an officially supported OS with Pioneer drivers - Audacity or Mixxx see the channels, but there is no signal. It probably works only with Pioneer proprietary application. But I hope that even mere output support will be useful for someone. (it is usable for playback, just not for DVS) Franta Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db..dcaf9eed9a41 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC

4 6

[PATCH] ALSA: pcm: oss: Fix regression by buffer overflow fix (again)
by Takashi Iwai 03 Apr '20

03 Apr '20

[ This is essentially the same fix as commit ae769d355664, but it's adapted to the latest code for 5.7; hence it contains no Fixes or other tags for avoid backport confusion -- tiwai ] The recent fix for the OOB access in PCM OSS plugins (commit f2ecf903ef06: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a regression on OSS applications. The patch introduced the size check in client and slave size calculations to limit to each plugin's buffer size, but I overlooked that some code paths call those without allocating the buffer but just for estimation. This patch fixes the bug by skipping the size check for those code paths while keeping checking in the actual transfer calls. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index fbda4ebf38b3..59d62f05658f 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -197,7 +197,8 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) } static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, - snd_pcm_sframes_t frames) + snd_pcm_sframes_t frames, + bool check_size) { struct snd_pcm_plugin *plugin, *plugin_next; @@ -209,7 +210,7 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, if (frames < 0) return frames; } - if (frames > plugin->buf_frames) + if (check_size && frames > plugin->buf_frames) frames = plugin->buf_frames; plugin = plugin_next; } @@ -217,13 +218,14 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, } static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, - snd_pcm_sframes_t frames) + snd_pcm_sframes_t frames, + bool check_size) { struct snd_pcm_plugin *plugin, *plugin_prev; plugin = snd_pcm_plug_last(plug); while (plugin && frames > 0) { - if (frames > plugin->buf_frames) + if (check_size && frames > plugin->buf_frames) frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { @@ -242,9 +244,9 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p return -ENXIO; switch (snd_pcm_plug_stream(plug)) { case SNDRV_PCM_STREAM_PLAYBACK: - return calc_src_frames(plug, drv_frames); + return calc_src_frames(plug, drv_frames, false); case SNDRV_PCM_STREAM_CAPTURE: - return calc_dst_frames(plug, drv_frames); + return calc_dst_frames(plug, drv_frames, false); default: snd_BUG(); return -EINVAL; @@ -257,9 +259,9 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc return -ENXIO; switch (snd_pcm_plug_stream(plug)) { case SNDRV_PCM_STREAM_PLAYBACK: - return calc_dst_frames(plug, clt_frames); + return calc_dst_frames(plug, clt_frames, false); case SNDRV_PCM_STREAM_CAPTURE: - return calc_src_frames(plug, clt_frames); + return calc_src_frames(plug, clt_frames, false); default: snd_BUG(); return -EINVAL; @@ -622,7 +624,7 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st src_channels = dst_channels; plugin = next; } - return snd_pcm_plug_client_size(plug, frames); + return calc_src_frames(plug, frames, true); } snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *dst_channels_final, snd_pcm_uframes_t size) @@ -632,7 +634,7 @@ snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, str snd_pcm_sframes_t frames = size; int err; - frames = snd_pcm_plug_slave_size(plug, frames); + frames = calc_src_frames(plug, frames, true); if (frames < 0) return frames; -- 2.16.4

1 0

[PATCH] ALSA: pcm: oss: Fix regression by buffer overflow fix
by Takashi Iwai 03 Apr '20

03 Apr '20

The recent fix for the OOB access in PCM OSS plugins (commit f2ecf903ef06: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a regression on OSS applications. The patch introduced the size check in client and slave size calculations to limit to each plugin's buffer size, but I overlooked that some code paths call those without allocating the buffer but just for estimation. This patch fixes the bug by skipping the size check for those code paths while keeping checking in the actual transfer calls. Fixes: f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") Tested-and-reported-by: Jari Ruusu <jari.ruusu(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 752d078908e9..50c35ecc8953 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -196,7 +196,9 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) return 0; } -snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t drv_frames, + bool check_size) { struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; int stream; @@ -209,7 +211,7 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { - if (drv_frames > plugin->buf_frames) + if (check_size && drv_frames > plugin->buf_frames) drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) @@ -222,7 +224,7 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); - if (drv_frames > plugin->buf_frames) + if (check_size && drv_frames > plugin->buf_frames) drv_frames = plugin->buf_frames; plugin = plugin_next; } @@ -231,7 +233,9 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p return drv_frames; } -snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames) +static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t clt_frames, + bool check_size) { struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; snd_pcm_sframes_t frames; @@ -252,14 +256,14 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } - if (frames > plugin->buf_frames) + if (check_size && frames > plugin->buf_frames) frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { - if (frames > plugin->buf_frames) + if (check_size && frames > plugin->buf_frames) frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { @@ -274,6 +278,18 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc return frames; } +snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t drv_frames) +{ + return plug_client_size(plug, drv_frames, false); +} + +snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t clt_frames) +{ + return plug_slave_size(plug, clt_frames, false); +} + static int snd_pcm_plug_formats(const struct snd_mask *mask, snd_pcm_format_t format) { @@ -630,7 +646,7 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st src_channels = dst_channels; plugin = next; } - return snd_pcm_plug_client_size(plug, frames); + return plug_client_size(plug, frames, true); } snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *dst_channels_final, snd_pcm_uframes_t size) @@ -640,7 +656,7 @@ snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, str snd_pcm_sframes_t frames = size; int err; - frames = snd_pcm_plug_slave_size(plug, frames); + frames = plug_slave_size(plug, frames, true); if (frames < 0) return frames; -- 2.16.4

1 0

[PATCH v2 1/1] ASoC: soc-component: add snd_soc_component_ack()
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

Current ALSA SoC is directly using component->driver->ops->xxx, thus, it is deep nested, and makes code difficult to read, and is not good for encapsulation. This patch adds new snd_soc_component_ack() and use it. Ack callback is revived because some ASoC driver is using it. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- include/sound/soc-component.h | 1 + sound/soc/soc-component.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 21 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2eef525..50b079272b47 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -381,6 +381,7 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream, unsigned long offset); int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream); int snd_soc_pcm_component_new(struct snd_pcm *pcm); void snd_soc_pcm_component_free(struct snd_pcm *pcm); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 79ffc2820ba9..b4be91cb9fa0 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -526,6 +526,24 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, return -EINVAL; } +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + /* FIXME. it returns 1st ack now */ + if (component->driver->ops && + component->driver->ops->ack) + return component->driver->ops->ack(substream); + } + + return -EINVAL; +} + int snd_soc_pcm_component_new(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b600d3eaaf5c..c8a5a9a9d35f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2941,6 +2941,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (!ops) continue; + if (ops->ack) + rtd->ops.ack = snd_soc_pcm_component_ack; if (ops->copy_user) rtd->ops.copy_user = snd_soc_pcm_component_copy_user; if (ops->page) base-commit: ef8e14794308a428b194f8b06ad9ae06b43466e4 -- 2.21.0

1 0

[PATCH 1/1] ASoC: soc-component: add snd_soc_component_ack()
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

Current ALSA SoC is directly using component->driver->ops->xxx, thus, it is deep nested, and makes code difficult to read, and is not good for encapsulation. This patch adds new snd_soc_component_ack() and use it. Ack callback is revived because some ASoC driver is using it. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- include/sound/soc-component.h | 3 +++ sound/soc/soc-component.c | 14 ++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 19 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 154d02fbbfed..a8c60efac8f7 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -104,6 +104,8 @@ struct snd_soc_component_driver { struct page *(*page)(struct snd_soc_component *component, struct snd_pcm_substream *substream, unsigned long offset); + int (*ack) (struct snd_soc_component *component, + struct snd_pcm_substream *substream); int (*mmap)(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct vm_area_struct *vma); @@ -415,6 +417,7 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream, unsigned long offset); int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream); int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd); void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 785a0385cc7f..b4b3245ef016 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -516,6 +516,20 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, return -EINVAL; } +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + int i; + + /* FIXME. it returns 1st ack now */ + for_each_rtd_components(rtd, i, component) + if (component->driver->ack) + return component->driver->ack(component, substream); + + return -EINVAL; +} + int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 289aebc15529..c6fddd1cb5aa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -3027,6 +3027,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.ioctl = snd_soc_pcm_component_ioctl; if (drv->sync_stop) rtd->ops.sync_stop = snd_soc_pcm_component_sync_stop; + if (drv->ack) + rtd->ops.ack = snd_soc_pcm_component_ack; if (drv->copy_user) rtd->ops.copy_user = snd_soc_pcm_component_copy_user; if (drv->page) -- 2.21.0

1 0

[PATCH 1/1] ASoC: soc-component: add snd_soc_component_ack()
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

Current ALSA SoC is directly using component->driver->ops->xxx, thus, it is deep nested, and makes code difficult to read, and is not good for encapsulation. This patch adds new snd_soc_component_ack() and use it. Ack callback is revived because some ASoC driver is using it. Change-Id: I85573cb43f3ca050c47ca9e3af3cfa9ec1fae3d0 Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- include/sound/soc-component.h | 1 + sound/soc/soc-component.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 21 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2eef525..50b079272b47 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -381,6 +381,7 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream, unsigned long offset); int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream); int snd_soc_pcm_component_new(struct snd_pcm *pcm); void snd_soc_pcm_component_free(struct snd_pcm *pcm); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 79ffc2820ba9..b4be91cb9fa0 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -526,6 +526,24 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, return -EINVAL; } +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + /* FIXME. it returns 1st ack now */ + if (component->driver->ops && + component->driver->ops->ack) + return component->driver->ops->ack(substream); + } + + return -EINVAL; +} + int snd_soc_pcm_component_new(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b600d3eaaf5c..c8a5a9a9d35f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2941,6 +2941,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (!ops) continue; + if (ops->ack) + rtd->ops.ack = snd_soc_pcm_component_ack; if (ops->copy_user) rtd->ops.copy_user = snd_soc_pcm_component_copy_user; if (ops->page) -- 2.21.0

1 0

[PATCH 0/1] reviving ack callback for 5.4
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

ACK callback is useful to notify application pointer in DMA buffer to DSP, because it is called after application pointer is updated in ALSA. DSP can use the pointer to control preprocessing, detect underrun and so on. I'd like to revive it from 5.4 if it is possible. Gyeongtaek Lee (1): ASoC: soc-component: add snd_soc_component_ack() include/sound/soc-component.h | 1 + sound/soc/soc-component.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 21 insertions(+) base-commit: ef8e14794308a428b194f8b06ad9ae06b43466e4 -- 2.21.0

1 0

[PATCH] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Hans de Goede 02 Apr '20

02 Apr '20

The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b..9c3bbf1df93e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7299,6 +7299,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.26.0

3 2

sof-hda-dsp: microphone device named "Headphones2"
by GitHub issues - opened 02 Apr '20

02 Apr '20

alsa-project/alsa-ucm-conf issue #19 was opened from tanuk: I was looking at the sof-hda-dsp UCM config, and I noticed that it defines a device named "Headphones2" which seems to actually represent a microphone. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/19 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: Intel: Multiple I/O PCM format support for pipe
by Mateusz Gorski 02 Apr '20

02 Apr '20

If a pipe supports multiple input/output formats, kcontrol is created and selection of pipe input and output configuration is done based on control set. Effectively allows user to select configuration using amixer settings. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> Signed-off-by: S, Pavan K <pavan.k.s(a)intel.com> --- include/uapi/sound/skl-tplg-interface.h | 1 + sound/soc/intel/skylake/skl-topology.c | 110 ++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 1 + 3 files changed, 112 insertions(+) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index 9eee32f5e407..f2711186c81f 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -18,6 +18,7 @@ */ #define SKL_CONTROL_TYPE_BYTE_TLV 0x100 #define SKL_CONTROL_TYPE_MIC_SELECT 0x102 +#define SKL_CONTROL_TYPE_MULTI_IO_SELECT 0x103 #define HDA_SST_CFG_MAX 900 /* size of copier cfg*/ #define MAX_IN_QUEUE 8 diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 344b06df0e15..de60b6873666 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -578,6 +578,38 @@ static int skl_tplg_unload_pipe_modules(struct skl_dev *skl, return ret; } +static bool skl_tplg_is_multi_fmt(struct skl_dev *skl, struct skl_pipe *pipe) +{ + int i; + struct skl_pipe_fmt *cur_fmt; + struct skl_pipe_fmt *next_fmt; + + if (pipe->nr_cfgs <= 1) + return false; + + if (pipe->conn_type != SKL_PIPE_CONN_TYPE_FE) + return true; + + for (i = 0; i < pipe->nr_cfgs - 1; i++) { + if (pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) { + cur_fmt = &pipe->configs[i].out_fmt; + next_fmt = &pipe->configs[i+1].out_fmt; + } else { + cur_fmt = &pipe->configs[i].in_fmt; + next_fmt = &pipe->configs[i+1].in_fmt; + } + + if (!CHECK_HW_PARAMS(cur_fmt->channels, cur_fmt->freq, + cur_fmt->bps, + next_fmt->channels, + next_fmt->freq, + next_fmt->bps)) + return true; + } + + return false; +} + /* * Here, we select pipe format based on the pipe type and pipe * direction to determine the current config index for the pipeline. @@ -600,6 +632,14 @@ skl_tplg_get_pipe_config(struct skl_dev *skl, struct skl_module_cfg *mconfig) return 0; } + if (skl_tplg_is_multi_fmt(skl, pipe)) { + pipe->cur_config_idx = pipe->pipe_config_idx; + pipe->memory_pages = pconfig->mem_pages; + dev_dbg(skl->dev, "found pipe config idx:%d\n", + pipe->cur_config_idx); + return 0; + } + if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE) { dev_dbg(skl->dev, "No conn_type detected, take 0th config\n"); pipe->cur_config_idx = 0; @@ -1314,6 +1354,71 @@ static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w, return 0; } +static int skl_tplg_multi_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct hdac_bus *bus = snd_soc_component_get_drvdata(component); + struct skl_dev *skl = bus_to_skl(bus); + struct skl_pipeline *ppl; + struct skl_pipe *pipe = NULL; + u32 *pipe_id; + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + + if (!ec) + return -EINVAL; + + pipe_id = ec->dobj.private; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == *pipe_id) { + pipe = ppl->pipe; + break; + } + } + if (!pipe) + return -EIO; + + ucontrol->value.enumerated.item[0] = pipe->pipe_config_idx; + + return 0; +} + +static int skl_tplg_multi_config_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct hdac_bus *bus = snd_soc_component_get_drvdata(component); + struct skl_dev *skl = bus_to_skl(bus); + struct skl_pipeline *ppl; + struct skl_pipe *pipe = NULL; + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + u32 *pipe_id; + + if (!ec) + return -EINVAL; + + if (ucontrol->value.enumerated.item[0] > ec->items) + return -EINVAL; + + pipe_id = ec->dobj.private; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == *pipe_id) { + pipe = ppl->pipe; + break; + } + } + if (!pipe) + return -EIO; + + pipe->pipe_config_idx = ucontrol->value.enumerated.item[0]; + + return 0; +} + static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, unsigned int __user *data, unsigned int size) { @@ -1853,6 +1958,11 @@ static const struct snd_soc_tplg_kcontrol_ops skl_tplg_kcontrol_ops[] = { .get = skl_tplg_mic_control_get, .put = skl_tplg_mic_control_set, }, + { + .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT, + .get = skl_tplg_multi_config_get, + .put = skl_tplg_multi_config_set, + }, }; static int skl_tplg_fill_pipe_cfg(struct device *dev, diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index e967800dbb62..06576147cc29 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -306,6 +306,7 @@ struct skl_pipe { struct skl_path_config configs[SKL_MAX_PATH_CONFIGS]; struct list_head w_list; bool passthru; + u32 pipe_config_idx; }; enum skl_module_state { -- 2.17.1

2 1

[PATCH] ASoC: Intel: Skylake: Add alternative topology binary name
by Mateusz Gorski 02 Apr '20

02 Apr '20

This commit adds alternative topology binary file name based on used machine driver and fallback to use this name after failed attempt to load topology file with name based on NHLT. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- sound/soc/intel/skylake/skl-topology.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..344b06df0e15 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3565,8 +3565,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->i2s_dev->name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3587,9 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index -- 2.17.1

3 4

[PATCH] ALSA:atmel:Use platform_get_irq_optional()
by Tang Bin 02 Apr '20

02 Apr '20

In order to simply code,because platform_get_irq() already has dev_err() message. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/atmel/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index a1dce9725..25cfd5710 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -715,7 +715,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - irq = platform_get_irq(pdev, 0); + irq = platform_get_irq_optional(pdev, 0); if (irq < 0) { dev_dbg(&pdev->dev, "could not get irq: %d\n", irq); return irq; -- 2.20.1.windows.1

2 1

[PATCH] ASoC: dpcm: allow start or stop during pause for backend
by 이경택 01 Apr '20

01 Apr '20

soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca0..8f6f0ad50288 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.21.0

3 2

[PATCH -next] ASoC: rt5682: Fix build error without CONFIG_I2C
by YueHaibing 01 Apr '20

01 Apr '20

If I2C is n but SoundWire is m, building fails: sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class module_i2c_driver(rt5682_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/rt5682.c:3716:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration Guard this use #ifdef CONFIG_I2C. Fixes: 5549ea647997 ("ASoC: rt5682: fix unmet dependencies") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt5682.c | 110 +++++++++++++++++++------------------- 1 file changed, 56 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index c9268a230daa..ff3efad874a4 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -33,12 +33,6 @@ #include "rt5682.h" #include "rt5682-sdw.h" -static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { - "AVDD", - "MICVDD", - "VBAT", -}; - static const struct rt5682_platform_data i2s_default_platform_data = { .dmic1_data_pin = RT5682_DMIC1_DATA_GPIO2, .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, @@ -984,25 +978,6 @@ static irqreturn_t rt5682_irq(int irq, void *data) return IRQ_HANDLED; } -static void rt5682_jd_check_handler(struct work_struct *work) -{ - struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, - jd_check_work.work); - - if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) - & RT5682_JDH_RS_MASK) { - /* jack out */ - rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); - - snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, - SND_JACK_HEADSET | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3); - } else { - schedule_delayed_work(&rt5682->jd_check_work, 500); - } -} - static int rt5682_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { @@ -3230,35 +3205,6 @@ static const struct i2c_device_id rt5682_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); -static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) -{ - - device_property_read_u32(dev, "realtek,dmic1-data-pin", - &rt5682->pdata.dmic1_data_pin); - device_property_read_u32(dev, "realtek,dmic1-clk-pin", - &rt5682->pdata.dmic1_clk_pin); - device_property_read_u32(dev, "realtek,jd-src", - &rt5682->pdata.jd_src); - device_property_read_u32(dev, "realtek,btndet-delay", - &rt5682->pdata.btndet_delay); - device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", - &rt5682->pdata.dmic_clk_rate); - device_property_read_u32(dev, "realtek,dmic-delay-ms", - &rt5682->pdata.dmic_delay); - - rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, - "realtek,ldo1-en-gpios", 0); - - if (device_property_read_string_array(dev, "clock-output-names", - rt5682->pdata.dai_clk_names, - RT5682_DAI_NUM_CLKS) < 0) - dev_warn(dev, "Using default DAI clk names: %s, %s\n", - rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], - rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); - - return 0; -} - static void rt5682_calibrate(struct rt5682_priv *rt5682) { int value, count; @@ -3526,6 +3472,61 @@ int rt5682_io_init(struct device *dev, struct sdw_slave *slave) EXPORT_SYMBOL_GPL(rt5682_io_init); #endif +#ifdef CONFIG_I2C +static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", + "VBAT", +}; + +static void rt5682_jd_check_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, + jd_check_work.work); + + if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK) { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682->jd_check_work, 500); + } +} + +static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) +{ + + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682->pdata.jd_src); + device_property_read_u32(dev, "realtek,btndet-delay", + &rt5682->pdata.btndet_delay); + device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", + &rt5682->pdata.dmic_clk_rate); + device_property_read_u32(dev, "realtek,dmic-delay-ms", + &rt5682->pdata.dmic_delay); + + rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + if (device_property_read_string_array(dev, "clock-output-names", + rt5682->pdata.dai_clk_names, + RT5682_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], + rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + + return 0; +} + static int rt5682_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3714,6 +3715,7 @@ static struct i2c_driver rt5682_i2c_driver = { .id_table = rt5682_i2c_id, }; module_i2c_driver(rt5682_i2c_driver); +#endif MODULE_DESCRIPTION("ASoC RT5682 driver"); MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); -- 2.17.1

2 3

Applied "ASoC: topology: use name_prefix for new kcontrol" to the asoc tree
by Mark Brown 01 Apr '20

01 Apr '20

The patch ASoC: topology: use name_prefix for new kcontrol has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From abca9e4a04fbe9c6df4d48ca7517e1611812af25 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=EC=9D=B4=EA=B2=BD=ED=83=9D?= <gt82.lee(a)samsung.com> Date: Wed, 1 Apr 2020 18:05:24 +0900 Subject: [PATCH] ASoC: topology: use name_prefix for new kcontrol Current topology doesn't add prefix of component to new kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Link: https://lore.kernel.org/r/009b01d60804$ae25c2d0$0a714870$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1f81cd2d29cf..87f75edba3dc 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.20.1

1 0

alsa-gobject: add library to operate ALSA Sequencer character device without queue feature
by GitHub pull_request - opened 01 Apr '20

01 Apr '20

alsa-project/alsa-gobject pull request #10 was opened from takaswie: This request is to add a new library, libalsaseq, to operate ALSA Sequencer character device. In design of ALSA Sequencer, one character device is added to Linux system. Usually a special file, '/dev/snd/seq', is used for the character device. Userspace applications can operate ALSA Sequencer via the character device. The library supports functionalities of ALSA Sequencer but without queue-related ones. The library includes some GObject-derived object. ALSASeq.UserClient maintains file descriptor of ALSA Sequencer character device, thus it's the center of user operation. The other objects represent data structures in UAPI of ALSA Sequencer. * ALSASeqSystemInfo - struct snd_seq_system_info * ALSASeqClientInfo - struct snd_seq_client_info * ALSASeqClientPool - struct snd_seq_client_pool * ALSASeqPortInfo - struct snd_seq_port_info * ALSASeqSubscribeData - struct snd_seq_port_subscribe The library includes some GObject boxed object just to represent data structures. ALSASeq.Addr is a thin wrapper to struct snd_seq_addr, and ALSASeq.Tstamp is a thin wrapper to struct snd_seq_timestamp. For operations without instances of ALSASeq.UserClient, the library has some global APIs. ALSA Sequencer has some macros. They're represented as GObject enumerations and flags. * ALSASeq.ClientType - one of NO_CLIENT, USER_CLIENT, KERNEL_CLIENT * ALSASeq.FilterAttrFlag - SNDRV_SEQ_FILTER_XXX * ALSASeq.PortCapFlag - SNDRV_PORT_CAP_XXX * ALSASeq.PortAttrFlag - NDRV_SEQ_PORT_TYPE_XXX * ALSASeq.EventType - SNDRV_SEQ_EVENT_XXX * ALSASeq.EventTimestampMode - SNDRV_SEQ_TIME_STAMP_XXX * ALSASeq.EventTimeMode - SNDRV_SEQ_TIME_MODE_XXX * ALSASeq.EventLengthMode - SNDRV_SEQ_EVENT_LENGTH_XXX * ALSASeq.EventPriorityMode - SNDRV_SEQ_PRIORITY_XXX * ALSASeq.PortSubscribeFlag - SNDRV_SEQ_PORT_SUBS_XXX * ALSASeq.QuerySubscribeType - SNDRV_SEQ_QUERY_SUBS_XXX Some specific values for addressing are also represented by GObject enumerations and flags. * ALSASeq.SpecificAddress - SNDRV_SEQ_ADDRESS_XXX * ALSASeq.SpecificClientId - SNDRV_SEQ_CLIENT_XXX * ALSASeq.SpecificPortId - SNDRV_SEQ_PORT_SYSTEM_XXX * ALSASeq.SpecificQueueId - SNDRV_SEQ_QUEUE_XXX ALSA Sequencer has many types of event. The library has an GObject abstract object, ALSASeq.Event. Two objects are derived from it: * ALSASeq.EventFixed * ALSASeq.EventVariable The former has some ways to access to the data of event. The data is represented by GObject boxed object if it's complicated structure: * ALSASeq.EventDataConnect - struct snd_seq_connect * ALSASeq.EventDataNote - struct snd_seq_ev_note * ALSASeq.EventDataCtl - struct snd_seq_ev_ctrl * ALSASeq.EventDataResult - struct snd_seq_result * ALSASeq.EventDataQueue - struct snd_seq_ev_queue_control The latter is for events to have variable length of data blob. ALSASeq.UserClient has 'schedule_event' API to schedule arbitrary event. It also has 'handle-event' GObject signal for applications to handle asynchronous event as an instance of ALSASeq.EventFixed or ALSASeq.EventVariable. The library doesn't have support for below structures since they're useless practically: * struct snd_seq_event_bounce * struct snd_seq_running_info In my later work, queue functions will be added. Request URL : https://github.com/alsa-project/alsa-gobject/pull/10 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 01 Apr '20

01 Apr '20

Current topology doesn't add prefix of component to new kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

2 2

[PATCH] ASoC: topology: use name_prefix for new kcontrol
by 이경택 01 Apr '20

01 Apr '20

Current topology doesn't add prefix of component to new kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

1 0

[PATCH] pcm: add chmap option to route plugin
by Jonas Holmberg 31 Mar '20

31 Mar '20

Add a config definition "chmap" to override the channel maps in the same way as in the hw and null plugins. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_route.c | 57 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 47 insertions(+), 10 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index bbcc6118..d3e5f3ff 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -104,6 +104,7 @@ typedef struct { int schannels; snd_pcm_route_params_t params; snd_pcm_chmap_t *chmap; + snd_pcm_chmap_query_t **chmap_override; } snd_pcm_route_t; #endif /* DOC_HIDDEN */ @@ -441,6 +442,7 @@ static int snd_pcm_route_close(snd_pcm_t *pcm) free(params->dsts); } free(route->chmap); + snd_pcm_free_chmaps(route->chmap_override); return snd_pcm_generic_close(pcm); } @@ -634,6 +636,9 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) snd_pcm_chmap_t *map, *slave_map; unsigned int src, dst, nsrcs; + if (route->chmap_override) + return _snd_pcm_choose_fixed_chmap(pcm, route->chmap_override); + slave_map = snd_pcm_generic_get_chmap(pcm); if (!slave_map) return NULL; @@ -660,8 +665,14 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) static snd_pcm_chmap_query_t **snd_pcm_route_query_chmaps(snd_pcm_t *pcm) { + snd_pcm_route_t *route = pcm->private_data; snd_pcm_chmap_query_t **maps; - snd_pcm_chmap_t *map = snd_pcm_route_get_chmap(pcm); + snd_pcm_chmap_t *map; + + if (route->chmap_override) + return _snd_pcm_copy_chmap_query(route->chmap_override); + + map = snd_pcm_route_get_chmap(pcm); if (!map) return NULL; maps = _snd_pcm_make_single_query_chmaps(map); @@ -818,10 +829,10 @@ err: return -EINVAL; } -static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, +static int find_matching_chmap(snd_pcm_chmap_query_t **chmaps, + snd_pcm_chmap_t *tt_chmap, snd_pcm_chmap_t **found_chmap, int *schannels) { - snd_pcm_chmap_query_t** chmaps = snd_pcm_query_chmaps(spcm); int i; *found_chmap = NULL; @@ -854,7 +865,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, int size = sizeof(snd_pcm_chmap_t) + c->channels * sizeof(unsigned int); *found_chmap = malloc(size); if (!*found_chmap) { - snd_pcm_free_chmaps(chmaps); return -ENOMEM; } memcpy(*found_chmap, c, size); @@ -863,8 +873,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, } } - snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { SNDERR("Found no matching channel map"); return -EINVAL; @@ -1252,6 +1260,7 @@ pcm.name { SCHANNEL REAL # route value (0.0 - 1.0) } } + [chmap MAP] # Override channel maps; MAP is a string array } \endcode @@ -1292,6 +1301,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, snd_pcm_route_ttable_entry_t *ttable = NULL; unsigned int csize, ssize; unsigned int cused, sused; + snd_pcm_chmap_query_t **chmaps = NULL; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; @@ -1306,31 +1316,45 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (strcmp(id, "ttable") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } tt = n; continue; } + if (strcmp(id, "chmap") == 0) { + chmaps = _snd_pcm_parse_config_chmaps(n); + if (!chmaps) { + SNDERR("Invalid channel map for %s", id); + return -EINVAL; + } + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } if (!slave) { SNDERR("slave is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } if (!tt) { SNDERR("ttable is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } err = snd_pcm_slave_conf(root, slave, &sconf, 2, SND_PCM_HW_PARAM_FORMAT, 0, &sformat, SND_PCM_HW_PARAM_CHANNELS, 0, &schannels); - if (err < 0) + if (err < 0) { + snd_pcm_free_chmaps(chmaps); return err; + } if (sformat != SND_PCM_FORMAT_UNKNOWN && snd_pcm_format_linear(sformat) != 1) { snd_config_delete(sconf); SNDERR("slave format is not linear"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } @@ -1345,13 +1369,19 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(tt_chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); return err; } if (tt_chmap) { - err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); + if (!chmaps) + chmaps = snd_pcm_query_chmaps(spcm); + if (chmaps) + err = find_matching_chmap(chmaps, tt_chmap, &chmap, + &schannels); free(tt_chmap); - if (err < 0) { + if (chmaps && err < 0) { + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1360,12 +1390,14 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } ttable = malloc(csize * ssize * sizeof(snd_pcm_route_ttable_entry_t)); if (ttable == NULL) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return -ENOMEM; } @@ -1374,6 +1406,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1385,9 +1418,13 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, free(ttable); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); } else { - ((snd_pcm_route_t*) (*pcmp)->private_data)->chmap = chmap; + snd_pcm_route_t *route = (*pcmp)->private_data; + + route->chmap = chmap; + route->chmap_override = chmaps; } return err; -- 2.24.1

2 1

[PATCH v2 1/3] dt-bindings: sound: convert rockchip spdif bindings to yaml
by Johan Jonker 31 Mar '20

31 Mar '20

Current dts files with 'spdif' nodes are manually verified. In order to automate this process rockchip-spdif.txt has to be converted to yaml. Also rk3188.dtsi, rk3288.dtsi use an extra fallback string, so change this in the documentation. Changed: "rockchip,rk3188-spdif", "rockchip,rk3066-spdif" "rockchip,rk3288-spdif", "rockchip,rk3066-spdif" Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- Changed V2: dmas and dma-names layout --- .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ----------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 94 ++++++++++++++++++++++ 2 files changed, 94 insertions(+), 45 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt deleted file mode 100644 index ec20c1271..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Rockchip SPDIF transceiver - -The S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-spdif" - - "rockchip,rk3188-spdif" - - "rockchip,rk3228-spdif" - - "rockchip,rk3288-spdif" - - "rockchip,rk3328-spdif" - - "rockchip,rk3366-spdif" - - "rockchip,rk3368-spdif" - - "rockchip,rk3399-spdif" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the SPDIF interrupt. -- dmas: DMA specifiers for tx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should be "tx" -- clocks: a list of phandle + clock-specifier pairs, one for each entry - in clock-names. -- clock-names: should contain following: - - "hclk": clock for SPDIF controller - - "mclk" : clock for SPDIF bus - -Required properties on RK3288: - - rockchip,grf: the phandle of the syscon node for the general register - file (GRF) - -Example for the rk3188 SPDIF controller: - -spdif: spdif@1011e000 { - compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; - reg = <0x1011e000 0x2000>; - interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dmac1_s 8>; - dma-names = "tx"; - clock-names = "hclk", "mclk"; - clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..d1c72c8a5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + maxItems: 1 + + dma-names: + const: tx + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru-common.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + }; -- 2.11.0

4 9

[PATCH] ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag
by Stephan Gerhold 31 Mar '20

31 Mar '20

At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- I'm still quite confused about the true meaning of SNDRV_PCM_INFO_BATCH, so please correct me if I'm wrong :) The tsched=0 workaround can be found in Linaro distributions for QCOM devices for example: - https://git.linaro.org/ci/fai.git/commit/?id=63494268b654d80df033f4cdeccf8f… - https://github.com/ndechesne/meta-qcom/commit/7035dfeadd1c434fc7613730ac380… This patch allows removing that workaround since audio then works without any configuration changes. --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index f6c7cddf08e8..125af00bba53 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -78,7 +78,7 @@ struct q6asm_dai_data { }; static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -100,7 +100,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { }; static struct snd_pcm_hardware q6asm_dai_hardware_playback = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | -- 2.26.0

5 4

[PATCH v2 1/3] ASoC: dapm: connect virtual mux with default value
by 이경택 31 Mar '20

31 Mar '20

Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial state. This is to ensure that the default mixer choice will be correctly powered up during initialization. Invert flag is used to select initial state of the virtual switch. Since actual hardware can't be disconnected by virtual switch, connected is better choice as initial state in many cases. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dapm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fb54e6fe254..17962564866d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -802,7 +802,13 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i, val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial state. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + p->connect = invert; } } -- 2.21.0

2 1

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

2 1

[PATCH] pcm: add chmap option to route plugin
by jonashg＠axis.com 31 Mar '20

31 Mar '20

From: Jonas Holmberg <jonashg(a)axis.com> Add a config definition "chmap" to override the channel maps in the same way as in the hw and null plugins. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_route.c | 57 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 47 insertions(+), 10 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index bbcc6118..a7b6552c 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -104,6 +104,7 @@ typedef struct { int schannels; snd_pcm_route_params_t params; snd_pcm_chmap_t *chmap; + snd_pcm_chmap_query_t **chmap_override; } snd_pcm_route_t; #endif /* DOC_HIDDEN */ @@ -441,6 +442,7 @@ static int snd_pcm_route_close(snd_pcm_t *pcm) free(params->dsts); } free(route->chmap); + snd_pcm_free_chmaps(route->chmap_override); return snd_pcm_generic_close(pcm); } @@ -634,6 +636,9 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) snd_pcm_chmap_t *map, *slave_map; unsigned int src, dst, nsrcs; + if (route->chmap_override) + return _snd_pcm_choose_fixed_chmap(pcm, route->chmap_override); + slave_map = snd_pcm_generic_get_chmap(pcm); if (!slave_map) return NULL; @@ -660,8 +665,14 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) static snd_pcm_chmap_query_t **snd_pcm_route_query_chmaps(snd_pcm_t *pcm) { + snd_pcm_route_t *route = pcm->private_data; snd_pcm_chmap_query_t **maps; - snd_pcm_chmap_t *map = snd_pcm_route_get_chmap(pcm); + snd_pcm_chmap_t *map; + + if (route->chmap_override) + return _snd_pcm_copy_chmap_query(route->chmap_override); + + map = snd_pcm_route_get_chmap(pcm); if (!map) return NULL; maps = _snd_pcm_make_single_query_chmaps(map); @@ -818,10 +829,10 @@ err: return -EINVAL; } -static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, +static int find_matching_chmap(snd_pcm_chmap_query_t **chmaps, + snd_pcm_chmap_t *tt_chmap, snd_pcm_chmap_t **found_chmap, int *schannels) { - snd_pcm_chmap_query_t** chmaps = snd_pcm_query_chmaps(spcm); int i; *found_chmap = NULL; @@ -854,7 +865,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, int size = sizeof(snd_pcm_chmap_t) + c->channels * sizeof(unsigned int); *found_chmap = malloc(size); if (!*found_chmap) { - snd_pcm_free_chmaps(chmaps); return -ENOMEM; } memcpy(*found_chmap, c, size); @@ -863,8 +873,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, } } - snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { SNDERR("Found no matching channel map"); return -EINVAL; @@ -1252,6 +1260,7 @@ pcm.name { SCHANNEL REAL # route value (0.0 - 1.0) } } + [chmap MAP] # Override channel maps; MAP is a string array } \endcode @@ -1292,6 +1301,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, snd_pcm_route_ttable_entry_t *ttable = NULL; unsigned int csize, ssize; unsigned int cused, sused; + snd_pcm_chmap_query_t **chmaps = NULL; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; @@ -1306,31 +1316,45 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (strcmp(id, "ttable") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } tt = n; continue; } + if (strcmp(id, "chmap") == 0) { + chmaps = _snd_pcm_parse_config_chmaps(n); + if (!chmaps) { + SNDERR("Invalid channel map for %s", id); + return -EINVAL; + } + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } if (!slave) { SNDERR("slave is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } if (!tt) { SNDERR("ttable is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } err = snd_pcm_slave_conf(root, slave, &sconf, 2, SND_PCM_HW_PARAM_FORMAT, 0, &sformat, SND_PCM_HW_PARAM_CHANNELS, 0, &schannels); - if (err < 0) + if (err < 0) { + snd_pcm_free_chmaps(chmaps); return err; + } if (sformat != SND_PCM_FORMAT_UNKNOWN && snd_pcm_format_linear(sformat) != 1) { snd_config_delete(sconf); SNDERR("slave format is not linear"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } @@ -1345,13 +1369,19 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(tt_chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); return err; } if (tt_chmap) { - err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); + if (!chmaps) + chmaps = snd_pcm_query_chmaps(spcm); + if (chmaps) + err = find_matching_chmap(chmaps, tt_chmap, &chmap, + &schannels); free(tt_chmap); - if (err < 0) { + if (chmaps && err < 0) { + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1360,12 +1390,14 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } ttable = malloc(csize * ssize * sizeof(snd_pcm_route_ttable_entry_t)); if (ttable == NULL) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return -ENOMEM; } @@ -1374,6 +1406,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1385,9 +1418,13 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, free(ttable); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); } else { - ((snd_pcm_route_t*) (*pcmp)->private_data)->chmap = chmap; + snd_pcm_route_t *route = (*pcmp)->private_data; + + route->chmap = chmap; + route->chmap_override = chmaps; } return err; -- 2.24.1

2 1

[PATCH] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 31 Mar '20

31 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 78def224f59c (ALSA: hda/realtek - Add Headset Mic supported) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

2 3

[PATCH 0/3] Properly fix headphone noise on the XPS 13 and other ALC256 devices
by Thomas Hebb 31 Mar '20

31 Mar '20

The root cause of various pervasive audio problems on the XPS 13 9350/9360, mostly relating to the headphone jack, turns out to be an undocumented feature of the ALC256 and similar codecs that routes audio along paths not exposed in the HDA node graph. The best we've had so far to configure this feature is magic numbers provided by Realtek, none of which have fully fixed all issues. This series documents the "PC Beep Hidden Register", which controls the feature and which I've reverse engineered using black box techniques, and uses my findings to hopefully fix the headphone issues on my XPS 13 once and for all. Thomas Hebb (3): ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 2 - .../sound/hd-audio/realtek-pc-beep.rst | 129 ++++++++++++++++++ sound/pci/hda/patch_realtek.c | 49 ++----- 4 files changed, 139 insertions(+), 42 deletions(-) create mode 100644 Documentation/sound/hd-audio/realtek-pc-beep.rst -- 2.25.2

3 6

[PATCH v2 0/3] Properly fix headphone noise on the XPS 13 and other ALC256 devices
by Thomas Hebb 31 Mar '20

31 Mar '20

The root cause of various pervasive audio problems on the XPS 13 9350/9360, mostly relating to the headphone jack, turns out to be an undocumented feature of the ALC256 and similar codecs that routes audio along paths not exposed in the HDA node graph. The best we've had so far to configure this feature is magic numbers provided by Realtek, none of which have fully fixed all issues. This series documents the "PC Beep Hidden Register", which controls the feature and which I've reverse engineered using black box techniques, and uses my findings to hopefully fix the headphone issues on my XPS 13 once and for all. Changes in v2: - Change fixed value from 0x4727 to 0x5757, which should behave identically, on advice from Kailang. Thomas Hebb (3): ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 2 - .../sound/hd-audio/realtek-pc-beep.rst | 129 ++++++++++++++++++ sound/pci/hda/patch_realtek.c | 49 ++----- 4 files changed, 139 insertions(+), 42 deletions(-) create mode 100644 Documentation/sound/hd-audio/realtek-pc-beep.rst -- 2.25.2

3 6

[PATCH 0/5] Add support for SOF on i.MX8M
by Daniel Baluta 31 Mar '20

31 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for SOF on i.MX8M family. First board from this family that has a DSP is i.MX8MP. First 2 patches are trying to fix some compilation issues, the next two are adding the imx8m support and the last one adds the devicetree binding. Daniel Baluta (4): ASoC: SOF: imx: fix undefined reference issue ASoC: SOF: imx: Add i.MX8M HW support ASoC: SOF: Add i.MX8MP device descriptor dt-bindings: dsp: fsl: Add fsl,imx8mp-dsp entry YueHaibing (1): ASoC: SOF: imx8: Fix randbuild error .../devicetree/bindings/dsp/fsl,dsp.yaml | 2 + sound/soc/sof/imx/Kconfig | 32 +- sound/soc/sof/imx/Makefile | 2 + sound/soc/sof/imx/imx8m.c | 279 ++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 14 + 5 files changed, 325 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/imx/imx8m.c -- 2.17.1

4 9

PATCH: ALSA USB MIDI: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Andreas Steinmetz 31 Mar '20

31 Mar '20

The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sourd/usb/midi.c 2020-03-26 17:34:40.073929497 +0100 +++ b/sound/usb/midi.c 2020-03-26 17:43:39.123252612 +0100 @@ -1839,7 +1839,8 @@ struct usb_host_endpoint *hostep; struct usb_endpoint_descriptor *ep; struct usb_ms_endpoint_descriptor *ms_ep; - int i, epidx; + int i, epidx, extralen; + unsigned char *extra; intf = umidi->iface; if (!intf) @@ -1863,12 +1864,21 @@ ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || + extralen = hostep->extralen; + extra = hostep->extra; +repeat: ms_ep = (struct usb_ms_endpoint_descriptor *)extra; + if (extralen < 1) + continue; + if (extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) - continue; + ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) { + if (!extra[0]) + continue; + extralen -= extra[0]; + extra += extra[0]; + goto repeat; + } if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) {

2 3

[PATCH] ASoC: snd_soc_dai_set_fmt add substream independence.
by Matt Flax 31 Mar '20

31 Mar '20

This patch is aims to start a stronger discussion on allowing both CPU and Codec dais to set formats independently based on direction. Currently it is very difficult to change stream formats because there is no way to specify it independently. A previous discussion w.r.t. a codec's stream dependent format setting required a codec's dai to be split. See here : https://mailman.alsa-project.org/pipermail/alsa-devel/2020-March/164492.html A previous discussion w.r.t. changing format based on the stream direction w.r.t. a CPU's dai showed the difficulty in setting fmt based on direection in thw hw_params function. See here : https://mailman.alsa-project.org/pipermail/alsa-devel/2020-March/165101.html It seems that it may be necessary to allow the stream direction as an input argument to snd_soc_dai_set_fmt. This patch is not complete, as there are many codecs and cpus which would require this change. However this patch is aimed as a discussion point. One example of a sound card which requires independent stream formats is an isolated sound card, such as the Audio Injector Isolated sound card. The magnetic isolation chips on the sound card require stream fomats to be different because of digital latency variations on the I2S lines. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- include/sound/soc-dai.h | 4 ++-- sound/soc/soc-core.c | 5 +++-- sound/soc/soc-dai.c | 4 ++-- 3 files changed, 7 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index eaaeb00e9e84..5071e006389b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -123,7 +123,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, int stream); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); @@ -186,7 +186,7 @@ struct snd_soc_dai_ops { * DAI format configuration * Called by soc_card drivers, normally in their hw_params. */ - int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt, int stream); int (*xlate_tdm_slot_mask)(unsigned int slots, unsigned int *tx_mask, unsigned int *rx_mask); int (*set_tdm_slot)(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a..27a6f01dc2d3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1474,10 +1474,11 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; unsigned int i; - int ret; + int ret, stream; for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + int stream = 0; + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt, stream); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, "ASoC: Failed to set DAI format: %d\n", ret); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 51031e330179..1f8ee7875656 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -94,11 +94,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); * * Configures the DAI hardware format and clocking. */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, int stream) { if (dai->driver->ops->set_fmt == NULL) return -ENOTSUPP; - return dai->driver->ops->set_fmt(dai, fmt); + return dai->driver->ops->set_fmt(dai, fmt, stream); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); -- 2.20.1

2 6

[PATCH 3/3] ASoC: fix regwmask
by 이경택 31 Mar '20

31 Mar '20

If regwshift is 32 and the selected architecture compiles '<<' operator for signed int literal into rotating shift, '1<<regwshift' became 1 and it makes regwmask to 0x0. The literal is set to unsigned long to get intended regwmask. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 652657dc6809..55ffb34be95e 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -825,7 +825,7 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long min = mc->min; @@ -874,7 +874,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long max = mc->max; -- 2.21.0

2 5

[PATCH v5 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 31 Mar '20

31 Mar '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: 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: rename asrc_priv to asrc 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 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 | 5 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 20 +- 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 | 2111 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3203 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

5 20

[PATCH] Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
by Takashi Iwai 31 Mar '20

31 Mar '20

This reverts commit 645c08f17f477915f6d900b767e789852f150054 which was reported to break the build a program using this header. The original issue was addressed in the alsa-lib side recently, so we can make the header more self-contained again. Reported-by: Dmitry V. Levin <ldv(a)altlinux.org> Fixes: 645c08f17f47 ("ALSA: uapi: Drop asound.h inclusion from asoc.h") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asoc.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6048553c119d..a74ca232f1fc 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -17,6 +17,7 @@ #define __LINUX_UAPI_SND_ASOC_H #include <linux/types.h> +#include <sound/asound.h> /* * Maximum number of channels topology kcontrol can represent. -- 2.16.4

1 0

[PATCH 1/3] ASoC: dapm: connect virtual mux with default value
by 이경택 31 Mar '20

31 Mar '20

Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial value 0. This is to ensure that the default mixer choice will be correctly powered up during initialization. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dapm.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fb54e6fe254..fc55a0534252 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -802,7 +802,16 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i, val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial value 0. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + val = 0; + if (invert) + val = max - val; + p->connect = !!val; } } -- 2.21.0

3 7

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

1 0

[PATCH 0/3] fixes for topology
by 이경택 31 Mar '20

31 Mar '20

Hi, I'd like to submit couple of patches which will fix few problems in corner case scenarios using topology. Any feedback will be greatly appreciated. Lee Gyeongtaek Lee (3): ASoC: dapm: connect virtual mux with default value ASoC: topology: use name_prefix for new kcontrol ASoC: fix regwmask sound/soc/soc-dapm.c | 11 ++++++++++- sound/soc/soc-ops.c | 4 ++-- sound/soc/soc-topology.c | 2 +- 3 files changed, 13 insertions(+), 4 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 2

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

1 0

[PATCH v3] ASoC: dt-bindings: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 31 Mar '20

31 Mar '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. This patch is assuming that "sound-dai" is already defined. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - uses maxItems for many place which were already defined. - assuming that "sound-dai" is already defined - (@.*) -> (@[0-9a-f]+) .../devicetree/bindings/sound/simple-card.txt | 351 ------------- .../bindings/sound/simple-card.yaml | 487 ++++++++++++++++++ 2 files changed, 487 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6e37b..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000000..cd5ea50daa22 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,487 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - maxItems: 1 + + bitclock-master: + description: Indicates dai-link bit clock master + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - maxItems: 1 + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + clocks: + maxItems: 1 + + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + label: + maxItems: 1 + + routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + maxItems: 1 + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + $ref: "#/definitions/dai-tdm-slot-num" + dai-tdm-slot-width: + $ref: "#/definitions/dai-tdm-slot-width" + clocks: + $ref: "#/definitions/clocks" + system-clock-frequency: + $ref: "#/definitions/system-clock-frequency" + system-clock-direction-out: + $ref: "#/definitions/system-clock-direction-out" + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": + const: 1 + "#size-cells": + const: 0 + + label: + $ref: "#/definitions/label" + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + +# use patternProperties to use "#definitions/xxx" +patternProperties: + "^simple-audio-card,widgets$": + $ref: "#/definitions/widgets" + "^simple-audio-card,routing$": + $ref: "#/definitions/routing" + "^simple-audio-card,cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" + "^simple-audio-card,hp-det-gpio$": + maxItems: 1 + "^simple-audio-card,mic-det-gpio$": + maxItems: 1 + + "^simple-audio-card,dai-link(@[0-9a-f]+)?$": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + maxItems: 1 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + + patternProperties: + "^cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.17.1

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[alsa-devel] [PATCH v3 0/7] ASoC: soc-pcm cleanup step2
by Kuninori Morimoto 31 Mar '20

31 Mar '20

Hi Mark These are v3 of soc-pcm cleanup. Original [5/7] [6/7] patch call open / start function for all DAI / component. And, it calls close / stop function for all if some of them got error. But in such case, error occured DAI / component don't need to call close / stop. This issue can be solved if it had flag, and my local tree has such patch. But it was planed to post to ML a little later. This time, I merged original patch and this new flag patch on [5/7] [6/7]. These mention that "it is prepare for soc-pcm-open() cleanup". But it will be happen later. These are based on v5.6-rc1 Kuninori Morimoto (8): 1) ASoC: soc-pcm: add snd_soc_runtime_action() 2) ASoC: soc-pcm: adjustment for DAI member 0 reset 3) ASoC: soc-pcm: add for_each_dapm_widgets() macro 4) ASoC: soc-pcm: don't use bit-OR'ed error 5) ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once 6) ASoC: soc-pcm: call snd_soc_component_open/close() once 7) ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() 8) ASoC: soc-pcm: tidyup soc_pcm_open() order include/sound/soc-component.h | 7 +- include/sound/soc-dai.h | 5 +- include/sound/soc-dapm.h | 5 + sound/soc/soc-component.c | 35 ++++-- sound/soc/soc-dai.c | 11 +- sound/soc/soc-dapm.c | 8 +- sound/soc/soc-pcm.c | 246 +++++++++++++++++++----------------------- 7 files changed, 161 insertions(+), 156 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/5] ALSA: Fix uapi headers
by Takashi Iwai 30 Mar '20

30 Mar '20

Hi, here is a series of fixes for include/uapi/sound/*.h files and the relevant code changes, so that they can be used for building alsa-tools programs again. The corresponding alsa-lib update will be followed. Takashi === Takashi Iwai (5): ALSA: emu10k1: Make uapi/emu10k1.h compilable again ALSA: hdsp: Make uapi/hdsp.h compilable again ALSA: hdspm: Drop linux/types.h inclusion in uapi header ALSA: uapi: Fix typos and header inclusion in asound.h ALSA: uapi: Drop asound.h inclusion from asoc.h include/uapi/sound/asoc.h | 1 - include/uapi/sound/asound.h | 6 +++--- include/uapi/sound/emu10k1.h | 38 ++++++++++++++++++++++++++------------ include/uapi/sound/hdsp.h | 4 +--- include/uapi/sound/hdspm.h | 2 -- sound/pci/emu10k1/emufx.c | 26 ++++++++++++++------------ sound/pci/rme9652/hdsp.c | 2 +- 7 files changed, 45 insertions(+), 34 deletions(-) -- 2.16.4

2 7

[PATCH] ASoC: soc-dai: fix DAI startup/shutdown sequence
by Pierre-Louis Bossart 30 Mar '20

30 Mar '20

The addition of a single flag to track the DAI status prevents the DAI startup sequence from being called on capture if the DAI is already used for playback. Fix by extending the existing code with one flag per direction. Reported-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Tested-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Fixes: b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc-dai.h | 2 +- sound/soc/soc-dai.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 78bac995db15..d4825b82c7a3 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -351,7 +351,7 @@ struct snd_soc_dai { /* bit field */ unsigned int probed:1; - unsigned int started:1; + unsigned int started[SNDRV_PCM_STREAM_LAST + 1]; }; static inline struct snd_soc_pcm_stream * diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 19142f6e533c..8f3cad8db89a 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,12 +295,12 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (!dai->started && + if (!dai->started[substream->stream] && dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); if (ret == 0) - dai->started = 1; + dai->started[substream->stream] = 1; return ret; } @@ -308,11 +308,11 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->started && + if (dai->started[substream->stream] && dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); - dai->started = 0; + dai->started[substream->stream] = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, -- 2.20.1

2 1

snd_hda_intel/sst-acpi sound breakage on suspend/resume since 5.6-rc1
by Dominik Brodowski 30 Mar '20

30 Mar '20

Hi! While 5.5.x works fine, mainline as of ac309e7744be (v5.6-rc6+) causes me some sound-related trouble: after boot, the sound works fine -- but once I suspend and resume my broadwell-based XPS13, I need to switch to headphone and back to speaker to hear something. But what I hear isn't music but garbled output. A few dmesg snippets from v5.6-rc6-9-gac309e7744be which might be of interest. I've highlighted the lines differing from v.5.5.x which might be of special interest: ... snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) usbcore: registered new interface driver snd-usb-audio snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops) input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input13 input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input14 input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input15 input: HDA Intel HDMI HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:03.0/sound/card0/input16 input: HDA Intel HDMI HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:03.0/sound/card0/input17 Console: switching to colour frame buffer device 240x67 !!! sst-acpi INT3438:00: WARN: Device release is not defined so it is not safe to unbind this driver while in use i915 0000:00:02.0: fb0: i915drmfb frame buffer device sst-acpi INT3438:00: DesignWare DMA Controller, 8 channels psmouse serio1: synaptics: Unable to query device: -5 haswell-pcm-audio haswell-pcm-audio: Direct firmware load for intel/IntcPP01.bin failed with error -2 haswell-pcm-audio haswell-pcm-audio: fw image intel/IntcPP01.bin not available(-2) haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 rt286 i2c-INT343A:00: ASoC: sink widget DMIC1 overwritten rt286 i2c-INT343A:00: ASoC: source widget DMIC1 overwritten broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> System Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Offload0 Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Offload1 Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Loopback Pin mapping ok broadwell-audio broadwell-audio: rt286-aif1 <-> snd-soc-dummy-dai mapping ok input: broadwell-rt286 Headset as /devices/pci0000:00/INT3438:00/broadwell-audio/sound/card1/input18 ... ALSA device list: #0: HDA Intel HDMI at 0xf7218000 irq 48 #1: DellInc.-XPS139343--0TM99H ... !!! haswell-pcm-audio haswell-pcm-audio: warning: stream is NULL, no stream to reset, ignore it. !!! haswell-pcm-audio haswell-pcm-audio: warning: stream is NULL, no stream to free, ignore it. (these last two messages already are printed a couple of time after boot, and then again during a suspend/resume cycle. On v.5.5.y, there are similar messages "no context buffer need to restore!"). Everything is built-in, no modules are loaded. Unfortunately, I cannot bisect this issue easily -- i915 was broken for quite some time on this system[*], prohibiting boot... Thanks for taking a look at this issue! Dominik [*] https://gitlab.freedesktop.org/drm/intel/issues/1151

8 38

[PATCH] dt-bindings: sound: cs42l51: Remove unneeded I2C unit name
by Fabio Estevam 30 Mar '20

30 Mar '20

The following warning is seen with 'make dt_binding_check': Documentation/devicetree/bindings/sound/cirrus,cs42l51.example.dts:18.15-34.11: Warning (unit_address_vs_reg): /example-0/i2c@0: node has a unit name, but no reg or ranges property Fix it by removing the unneeded i2c unit name. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml index efce847a3408..83f44f07ac3f 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -49,7 +49,7 @@ required: examples: - | #include <dt-bindings/gpio/gpio.h> - i2c@0 { + i2c { #address-cells = <1>; #size-cells = <0>; -- 2.17.1

3 2

[PATCH] ASoC: bcm: Fix pointer cast warning
by Takashi Iwai 30 Mar '20

30 Mar '20

The NULL check can be done gracefully without cast. It fixes a compile warning like: sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] Fixes: 88eb404ccc3e ("ASoC: brcm: Add DSL/PON SoC audio driver") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/bcm/bcm63xx-pcm-whistler.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index e46c390683e7..b7a1efc7406e 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -181,7 +181,7 @@ bcm63xx_pcm_pointer(struct snd_soc_component *component, snd_pcm_uframes_t x; struct bcm63xx_runtime_data *prtd = substream->runtime->private_data; - if ((void *)prtd->dma_addr_next == NULL) + if (!prtd->dma_addr_next) prtd->dma_addr_next = substream->runtime->dma_addr; x = bytes_to_frames(substream->runtime, -- 2.16.4

2 1

[PATCH] ASoC: AMD: Clear format bits before setting them
by Akshu Agrawal 30 Mar '20

30 Mar '20

This avoids residual bit form previous format when the format is changed. Hence, the resultant format is not an invalid one. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 1 + sound/soc/amd/raven/acp3x.h | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 3a3c47e820ab..f160d35a6832 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -139,6 +139,7 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg); } val = rv_readl(rtd->acp3x_base + reg_val); + val &= ~ACP3x_ITER_IRER_SAMP_LEN_MASK; val = val | (rtd->xfer_resolution << 3); rv_writel(val, rtd->acp3x_base + reg_val); return 0; diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 21e7ac017f2b..03fe93913e12 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -76,6 +76,8 @@ #define ACP_POWERED_OFF 0x02 #define ACP_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_ITER_IRER_SAMP_LEN_MASK 0x38 + struct acp3x_platform_info { u16 play_i2s_instance; u16 cap_i2s_instance; -- 2.20.1

2 1

[PATCH 0/4] ucm2: Add support to SDM845 based platforms
by Srinivas Kandagatla 30 Mar '20

30 Mar '20

This patchset adds UCM support for WSA881x, WCD934x codec and two platforms, DB845c DragonBoard and Lenovo YOGA C360 Laptop that use these codecs. Tested on Ubuntu 20.04 with pulseaudio 13 along with with this fix https://github.com/pulseaudio/pulseaudio/commit/cabd387e26b5fe03661f4b89482… Thanks, srini Srinivas Kandagatla (4): ucm2: codecs: wcd934x: Add Codec sequences. ucm2: codecs: wsa881x: add codec sequences ucm2: DB845c: Add ucm for DB845c board ucm2: Add ucm for Lenovo-YOGA-C630-13Q50 laptop ucm2/DB845c/DB845c.conf | 11 +++ ucm2/DB845c/HDMI.conf | 33 +++++++ ucm2/DB845c/HiFi.conf | 46 +++++++++ ucm2/Lenovo-YOGA-C630-13Q50/HiFi.conf | 98 +++++++++++++++++++ .../Lenovo-YOGA-C630-13Q50.conf | 6 ++ ucm2/codecs/wcd934x/DefaultDisableSeq.conf | 5 + ucm2/codecs/wcd934x/DefaultEnableSeq.conf | 14 +++ ucm2/codecs/wcd934x/HeadphoneDisableSeq.conf | 6 ++ ucm2/codecs/wcd934x/HeadphoneEnableSeq.conf | 6 ++ .../wcd934x/HeadphoneMicDisableSeq.conf | 2 + .../codecs/wcd934x/HeadphoneMicEnableSeq.conf | 3 + ucm2/codecs/wcd934x/SpeakerDisableSeq.conf | 4 + ucm2/codecs/wcd934x/SpeakerEnableSeq.conf | 4 + ucm2/codecs/wsa881x/DefaultEnableSeq.conf | 8 ++ ucm2/codecs/wsa881x/SpeakerDisableSeq.conf | 8 ++ ucm2/codecs/wsa881x/SpeakerEnableSeq.conf | 8 ++ 16 files changed, 262 insertions(+) create mode 100644 ucm2/DB845c/DB845c.conf create mode 100644 ucm2/DB845c/HDMI.conf create mode 100644 ucm2/DB845c/HiFi.conf create mode 100644 ucm2/Lenovo-YOGA-C630-13Q50/HiFi.conf create mode 100644 ucm2/Lenovo-YOGA-C630-13Q50/Lenovo-YOGA-C630-13Q50.conf create mode 100644 ucm2/codecs/wcd934x/DefaultDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/DefaultEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneMicDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneMicEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/SpeakerDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/SpeakerEnableSeq.conf create mode 100644 ucm2/codecs/wsa881x/DefaultEnableSeq.conf create mode 100644 ucm2/codecs/wsa881x/SpeakerDisableSeq.conf create mode 100644 ucm2/codecs/wsa881x/SpeakerEnableSeq.conf -- 2.21.0

2 7

[GIT PULL] ASoC updates for v5.7
by Mark Brown 30 Mar '20

30 Mar '20

The following changes since commit 16fbf79b0f83bc752cee8589279f1ebfe57b3b6e: Linux 5.6-rc7 (2020-03-22 18:31:56 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.7 for you to fetch changes up to 1c521d7e62262793789845989edca57dea24eb7d: Merge branch 'asoc-5.7' into asoc-next (2020-03-27 17:29:20 +0000) ---------------------------------------------------------------- ASoC: Updates for v5.7 This is a very big update for the core since Morimoto-san has been rather busy continuing his refactorings to clean up a lot of the cruft that we have accumilated over the years. We've also gained several new drivers, including initial (but still not complete) parts of the Intel SoundWire support. - Lots of refactorings to modernize the code from Morimoto-san. - Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven. - Continued refactoring and fixing of the Intel support. - Soundwire and more advanced clocking support for Realtek RT5682. - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and TLV320ADCX140. ---------------------------------------------------------------- Akshu Agrawal (4): ASoC: amd: Add machine driver for Raven based platform ASoC: amd: Allow I2S wake event after ACP is powerd On ASoc: amd: Add DMIC switch capability to machine driver ASoC: amd: Fix compile warning of argument type Amadeusz Sławiński (1): ASoC: topology: Perform component check upfront Amery Song (2): ASoC: SOF: Intel: hda: remove unnecessary ROM IPC filter function ASoC: SOF: Intel: remove unnecessary waitq before loading firmware Baolin Wang (1): ASoC: sprd: Allow the MCDT driver to build into modules Bard Liao (5): ASoC: Return error if the function does not support multi-cpu ASoC: pcm: check if cpu-dai supports a given stream ASoC: SOF: Intel: hda-dai: add stream capability ASoC: SOF: topology: connect dai widget to all cpu-dais ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Benjamin Gaignard (1): ASoC: Convert cirrus,cs42l51 to json-schema Cezary Rojewski (20): ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Expose SDnFMT helpers ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs ASoC: SOF: Fix probe point getter ASoC: Intel: Skylake: Remove superfluous chip initialization ASoC: Intel: Skylake: Select hda configuration permissively ASoC: Intel: Skylake: Enable codec wakeup during chip init ASoC: Intel: Skylake: Shield against no-NHLT configurations ASoC: Intel: Allow for ROM init retry on CNL platforms ASoC: Intel: Skylake: Await purge request ack on CNL ASoC: SOF: Intel: Fix stream cleanup on hw free ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components ASoC: Intel: bdw-rt5677: Revert SSP0 link to use dummy components ASoC: Intel: bdw-rt5650: Revert SSP0 link to use dummy components Charles Keepax (1): ASoC: samsung: Update dependencies for Arizona machine drivers Colin Ian King (6): ASoC: Intel: mrfld: return error codes when an error occurs ASoC: rt5659: remove redundant assignment to variable idx ASoC: ti: davinci-mcasp: remove redundant assignment to variable ret ASoC: Intel: mrfld: fix incorrect check on p->sink ASoC: mchp-i2s-mcc: make signed 1 bit bitfields unsigned ASoC: amd: acp3x-pcm-dma: clean up two indentation issues Curtis Malainey (1): ASoC: Intel: Make glk+rt5682 echo ref dynamic Dafna Hirschfeld (1): dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format. Dan Murphy (10): ASoC: tas2562: Add support for ISENSE and VSENSE dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family ASoC: tas2562: Add support for digital volume control ASoC: tlv320adcx140: Add DRE and AGC support ASoC: tlv320adcx140: Add decimation filter support ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding ASoC: tas2562: Add entries for the TAS2563 audio amplifier ASoC: tlv320adcx140: Fix mic_bias and vref device tree verification ASoC: tlv320adcx140: Remove undocumented property Derek Fang (3): ASoC: rl6231: Add new supports on rl6231 ASoC: rt5682: Add CCF usage for providing I2S clks ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage Dmitry Osipenko (2): ASoC: tegra: tegra_wm8903: Support DAPM events for built-in microphone ASoC: tegra-wm8903: Document built-in microphone audio source Enric Balletbo i Serra (1): ASoC: amd: AMD RV RT5682 should depends on CROS_EC Geert Uytterhoeven (7): spi: pxa2xx: Enable support for compile-testing ASoC: Use imply for SND_SOC_ALL_CODECS ASoC: sh: fsi: Restore devm_ioremap() alignment ASoC: Fix SND_SOC_ALL_CODECS imply SPI fallout ASoC: Fix SND_SOC_ALL_CODECS imply I2C fallout ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout Guennadi Liakhovetski (7): ASoC: (cosmetic) simplify dpcm_prune_paths() ASoC: export DPCM runtime update functions ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load() ASoC: SOF: fix uninitialised "work" with VirtIO Gustavo A. R. Silva (1): ASoC: wm0010: Replace zero-length array with flexible-array member Jack Yu (2): ASoC: rt1015: fix typo for bypass boost control ASoC: rt1015: modify some structure to be static. James Schulman (1): MAINTAINERS: Update Cirrus Logic codec driver maintainers Jaska Uimonen (1): ASoC: SOF: ipc: check ipc return value before data copy Jeff Chang (1): ASoC: MT6660 update to 1.0.8_G Jerome Brunet (23): ASoC: core: allow a dt node to provide several components ASoC: meson: g12a: extract codec-to-codec utils ASoC: meson: aiu: add audio output dt-bindings ASoC: meson: aiu: add i2s and spdif support ASoC: meson: aiu: add hdmi codec control support ASoC: meson: aiu: add internal dac codec control support ASoC: meson: axg: extract sound card utils ASoC: meson: gx: add sound card dt-binding documentation ASoC: meson: gx: add sound card support ASoC: core: ensure component names are unique ASoC: meson: aiu: remove unused encoder structure ASoC: meson: aiu: fix clk bulk size allocation ASoC: meson: aiu: fix irq registration ASoC: meson: aiu: fix acodec dai input name init ASoC: meson: codec-glue: fix pcm format cast warning ASoC: meson: aiu: simplify component addition ASoC: fix card registration regression. ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver ASoC: dpcm: remove confusing trace in dpcm_get_be() ASoC: meson: g12a: add toacodec dt-binding documentation ASoC: meson: g12a: add internal DAC glue driver ASoC: meson: axg-card: add toacodec support Johan Jonker (2): dt-bindings: sound: convert rockchip i2s bindings to yaml dt-bindings: sound: rockchip-i2s: add #sound-dai-cells property Jonghwan Choi (1): ASoC: tas2562: Fixed incorrect amp_level setting. Kai Vehmanen (10): ASoC: intel/skl/hda - add no-HDMI cases to generic HDA driver ASoC: SOF: Intel: hda: allow operation without i915 gfx ASoC: soc-pcm: fix state tracking error in snd_soc_component_open/close() MAINTAINERS: add entry for Sound Open Firmware drivers ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC ASoC: SOF: Intel: hda: do not leave clock gating off upon error ASoC: SOF: Intel: hda: call codec wake at chip init Karol Trzcinski (2): ASoC: SOF: Make sof_ipc_ext_data enum more rigid ASoC: SOF: Remove SOF_IPC_EXT_DMA_BUFFER Kevin Li (2): ASoC: brcm: Add DSL/PON SoC audio driver ASoC: brcm: DSL/PON SoC device tree bindings of audio driver Keyon Jie (2): ASoC: SOF: pcm: skip DMA buffer pre-allocation ASoC: SOF: Intel: hda-loader: clear the IPC ack bit after FW_PURGE done Kuninori Morimoto (95): ASoC: soc-pcm: add snd_soc_runtime_action() ASoC: soc-pcm: adjustment for DAI member 0 reset ASoC: soc-pcm: add for_each_dapm_widgets() macro ASoC: soc-pcm: don't use bit-OR'ed error ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once ASoC: soc-pcm: call snd_soc_component_open/close() once ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() ASoC: soc-pcm: tidyup soc_pcm_open() order ASoC: soc-pcm: move dai_get_widget() ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code ASoC: soc-pcm: use goto and remove multi return ASoC: soc-pcm: merge playback/cature_active into stream_active ALSA: pcm.h: add for_each_pcm_streams() ASoC: soc-core: use for_each_pcm_streams() macro ASoC: soc-pcm: use for_each_pcm_streams() macro ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro ASoC: sof: sof-audio: use for_each_pcm_streams() macro ALSA: usx2y: use for_each_pcm_streams() macro ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro ASoC: soc-pcm: add snd_soc_dai_get_pcm_stream() ASoC: soc-pcm: cleanup soc_pcm_apply_msb() ASoC: soc-pcm: add snd_soc_dai_get_widget() ASoC: soc-pcm: merge dpcm_run_new/old_update() into dpcm_fe_runtime_update() ASoC: soc-pcm: move dpcm_path_put() to soc-pcm.c ASoC: soc-pcm: move CONFIG_DEBUG_FS functions to top side ASoC: soc-pcm: add dpcm_create/remove_debugfs_state() ASoC: soc-dapm: don't use rtd->cpu_dai on for_each_rtd_cpu_dai() ASoC: soc-pcm: use defined stream ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() ASoC: soc-pcm: move dpcm_fe_dai_close() ASoC: soc-pcm: add dpcm_fe_dai_cleanup() ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() ASoC: soc-pcm: check DAI's activity more simply ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. ASoC: simple-card-utils: use for_each_pcm_streams() ASoC: soc.h: add for_each_rtd_codecs/cpus_dai() macro ASoC: Intel: use for_each_rtd_codecs/cpus_dai() macro ASoC: mediatek: use for_each_rtd_codecs/cpus_dai() macro ASoC: meson: use for_each_rtd_codecs/cpus_dai() macro ASoC: qcom: use for_each_rtd_codecs/cpus_dai() macro ASoC: soc: use for_each_rtd_codecs/cpus_dai() macro ASoC: soc.h: remove non plural form for_each_xxx macro ASoC: soc-dapm: add for_each_card_dapms() macro ASoC: soc-dapm: add for_each_card_widgets() macro ASoC: soc-core: Merge CPU/Codec DAIs ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-pcm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new() ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Lukas Bulwahn (1): ASoC: MT6660: make spdxcheck.py happy Marek Szyprowski (1): ASoC: samsung: Silence warnings during deferred probe Mark Brown (37): Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 ASoC: soc-pcm cleanup step4 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 SoundWire: ASoC interfaces for multi-cpu dais and DisCo helpers ASoC: Add Multi CPU DAI support Merge series "ASoC: SOF: updates for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ASoC: wcd934x: minor fixes" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "simple-audio-card codec2codec support" from Samuel Holland <samuel(a)sholland.org>: Merge series "ASoC: soc-pcm cleanup step5" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: soc-pcm cleanup step6" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: rt5682: fix Kconfig/compilation issues when I2C is not selected" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge tag 'v5.6-rc5' into asoc-5.7 Merge series "ASoC: qdsp6: fix default FE dais and routings." from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: SOF: preparatory patches" from Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com>: Merge series "ASoC: brcm: add dsl and pon chip audio driver" from Kevin Li <kevin-ke.li(a)broadcom.com>: Merge series "ASoC: Intel: machine driver updates for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: multi-cpu dais, IPC and Intel improvements for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ALSA: compress: Add wma, alac and ape support" from Vinod Koul <vkoul(a)kernel.org>: Merge series "ASoC: sdm845: fix soundwire stream handling" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: stm32: manage rebind issue" from Olivier Moysan <olivier.moysan(a)st.com>: Merge branch 'topic/ro_wordlength' of git://git.kernel.org/.../vkoul/soundwire into asoc-5.7 Merge series "Support built-in Mic on Tegra boards that use WM8903" from Dmitry Osipenko <digetx(a)gmail.com>: ASoC: pxa: Select regmap from AC'97 machines ASoC: pxa: Enable AC'97 bus support for PXA machines Merge series "ASoC: SOF: cleanups and improvements" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Intel: machine driver updates" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: rt1308-sdw: configure amplifier with set_tdm_slot()" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: Intel: add SoundWire support" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: remove rtd->cpu/codec_dai{s}" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ASoC: Intel: add SoundWire machine driver" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'asoc-5.6' into asoc-linus Merge branch 'asoc-5.7' into asoc-next Martin Blumenstingl (3): ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings ASoC: meson: aiu: introduce a struct for platform specific information ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families Nathan Chancellor (1): ASoC: wcd934x: Remove some unnecessary NULL checks Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Oder Chiou (8): ASoC: rt5682: Add the field "is_sdw" of private data ASoC: rt5682: Add the soundwire support ASoC: rt5682: Revise the function name ASoC: rt5682: Revise the DAC1 volume setting ASoC: rt5682: Fine tune the HP performance in soundwire mode ASoC: rt5682: Add a property for DMIC clock rate ASoC: rt5682: Add a property for DMIC delay ASoC: rt5682: Add the descriptions for the DMIC clock rate and delay settings Olivier Moysan (11): ASoC: stm32: sai: manage error when getting reset controller ASoC: stm32: spdifrx: manage error when getting reset controller ASoC: stm32: i2s: manage error when getting reset controller ASoC: stm32: sai: improve error management on probe deferral ASoC: stm32: spdifrx: improve error management on probe deferral ASoC: stm32: i2s: improve error management on probe deferral ASoC: dt-bindings: stm32: convert i2s to json-schema ASoC: dt-bindings: stm32: convert spdfirx to json-schema ASoC: stm32: spdifrx: fix regmap status check ASoC: stm32: spdifrx: manage rebind issue ASoC: stm32: i2s: manage rebind issue Paul Cercueil (6): ASoC: jz4740-i2s: Fix divider written at incorrect offset in register ASoC: jz4740-i2s: Add local dev variable in probe function ASoC: jz4740-i2s: Avoid passing enum as match data ASoC: jz4740-i2s: Add support for the JZ4760 ASoC: jz4740-i2s: Add support for the JZ4770 ASoC: Convert jz4740-i2s doc to YAML Paul Olaru (3): ASoC: SOF: Rename i.MX8 platform to i.MX8X ASoC: SOF: imx8: Add ops for i.MX8QM ASoC: SOF: Add i.MX8QM device descriptor Peter Ujfalusi (5): ASoC: ti: Add udma-pcm platform driver for UDMA ASoC: ti: davinci-mcasp: Add support for platforms using UDMA ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback ALSA: dmaengine_pcm: No need to take runtime reference twice in pcm_pointer Pierre-Louis Bossart (20): ASoC: Intel: sof_pcm512x: add support for SOF platforms with pcm512x ASoC: Intel: BXT: switch pcm512x based boards to sof_pcm512x ASoC: Intel: CHT: add support for pcm512x boards ASoC: soc-dai: add get_sdw_stream() callback ASoC: soc-core: disable route checks for legacy devices ASoC: Intel: skl_nau88l25_ssm4567: disable route checks ASoC: rt5682: fix unmet dependencies ASoC: rt5682-sdw: fix 'defined but not used' pm functions ASoC: Intel: don't use GFP_ATOMIC for machine driver contexts ASoC: rt1308-sdw: add set_tdm_slot() support ASoC: rt1308-sdw: use slot and rx_mask to configure stream ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (3): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms Ranjani Sridharan (11): ASoC: SOF: Do not reset hw_params for streams that ignored suspend ASoC: SOF: pm: Unify suspend/resume routines ASoC: SOF: Add system_suspend_target field to struct snd_sof_dev ASoC: SOF: pm: Introduce DSP power states ASoC: SOF: Move DSP power state transitions to platform-specific ops ASoC: SOF: audio: Add helper to check if only D0i3 streams are active ASoC: SOF: Intel: hda: Amend the DSP state transition diagram ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0 ASoC: SOF: Intel: hda: Allow trace DMA in S0 when DSP is in D0I3 for debug ASoC: SOF: Intel: hda: use snd_sof_dsp_set_power_state() op ASoC: SOF: Intel: hda: Improve DSP state logging Ravulapati Vishnu vardhan rao (1): ASoC: amd: Adding TDM support in hw_params. Rob Herring (1): ASoC: dt-bindings: google, cros-ec-codec: Fix dtc warnings in example Robin Murphy (2): ASoC: dt-bindings: Make RK3328 codec GPIO explicit ASoC: rockchip: Make RK3328 GPIO_MUTE control explicit Samuel Holland (4): ASoC: sun8i-codec: Remove unused dev from codec struct ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec2codec DAI links Sathyanarayana Nujella (2): ASoC: Intel: sof_rt5682: Add support for tgl-max98373-rt5682 ASoC: Intel: common: Add mach table for tgl-max98373-rt5682 Shreyas NC (4): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add dapm_add_valid_dai_widget helper ASoC: Add multiple CPU DAI support in DAPM Shuming Fan (3): ASoC: rt5682: fix the random recording noise of headset ASoC: rt5682: remove noisy debug messages ASoC: rt5682: move DAI clock registry to I2S mode Srinivas Kandagatla (10): ASoC: wcd934x: fix High Accuracy Buck enable ASoC: wcd934x: remove unused headers ASoC: wcd9335: fix address map representation ASoC: qdsp6: q6asm-dai: only enable dais from device tree ASoC: qdsp6: q6routing: remove default routing ASoC: codecs: wsa881x: request gpio direction before setting ASoC: soc-dai: return proper error for get_sdw_stream() ASoC: codecs: wsa881x: remove soundwire stream handling ASoC: qcom: sdm845: handle soundwire stream soundwire: stream: Add read_only_wordlength flag to port properties Stephan Gerhold (1): ASoC: soc-pcm: fix regression in soc_new_pcm() Takashi Iwai (2): ASoC: pcm: Fix (again) possible buffer overflow in dpcm state sysfs output ASoC: wm_adsp: Use scnprintf() for the limited buffer output Tomasz Lauda (1): ASoC: SOF: add core id to sof_ipc_comp Tzung-Bi Shih (12): ASoC: mediatek: mt8173-rt5650: support HDMI jack reporting drm/mediatek: exit earlier if failed to register audio driver drm/mediatek: support HDMI jack status reporting ASoC: max98357a: move control of SD_MODE to DAPM ASoC: mediatek: mt8183-da7219: add speaker switch ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1 ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out ASoC: mediatek: mt8183-da7219: apply some refactors drm/mediatek: fix race condition for HDMI jack status reporting ASoC: dapm: select sleep_state when initializing PINCTRL widget ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out Vinod Koul (9): ALSA: compress: add wma codec profiles ALSA: compress: Add wma decoder params ASoC: qcom: q6asm: pass codec profile to q6asm_open_write ASoC: qcom: q6asm: add support to wma config ASoC: qcom: q6asm-dai: add support to wma decoder ALSA: compress: add alac & ape decoder params ASoC: qcom: q6asm: add support for alac and ape configs ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders ALSA: compress: bump the version Wolfram Sang (1): ASoC: pxa: magician: convert to use i2c_new_client_device() Yong Zhi (4): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support ASoC: max98357a: Add ACPI HID MAX98360A ASoC: intel: sof_da7219_max98373: Add speaker switch Yu-Hsuan Hsu (1): ASoC: cros_ec_codec: Support setting bclk ratio YueHaibing (6): ASoC: Remove unused including <linux/version.h> ASoC: wcd934x: Remove set but not unused variable 'hph_comp_ctrl7' ASoC: rt5682: Make rt5682_clock_config static ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver ASoc: amd: acp3x: Add missing include <linux/io.h> ASoC: wm8974: remove unused variables derek.fang (1): ASoC: rt5682: Enable PLL2 function kbuild test robot (1): ASoC: meson: aiu: fix semicolon.cocci warnings tangbin (2): ASoC: zte: zx-spdif: remove redundant dev_err message ASoC: zte: zx-tdm: remove redundant variables dev .../devicetree/bindings/sound/amlogic,aiu.yaml | 113 ++ .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 + .../bindings/sound/amlogic,gx-sound-card.yaml | 113 ++ .../devicetree/bindings/sound/amlogic,t9015.yaml | 58 + .../bindings/sound/brcm,bcm63xx-audio.txt | 29 + .../devicetree/bindings/sound/cirrus,cs42l51.yaml | 69 + .../devicetree/bindings/sound/cs42l51.txt | 33 - .../bindings/sound/google,cros-ec-codec.txt | 44 - .../bindings/sound/google,cros-ec-codec.yaml | 67 + .../devicetree/bindings/sound/ingenic,aic.yaml | 92 ++ .../bindings/sound/ingenic,jz4740-i2s.txt | 23 - .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + .../bindings/sound/rockchip,rk3328-codec.txt | 7 +- .../devicetree/bindings/sound/rockchip-i2s.txt | 49 - .../devicetree/bindings/sound/rockchip-i2s.yaml | 111 ++ Documentation/devicetree/bindings/sound/rt5682.txt | 18 + .../devicetree/bindings/sound/st,stm32-i2s.txt | 62 - .../devicetree/bindings/sound/st,stm32-i2s.yaml | 87 ++ .../devicetree/bindings/sound/st,stm32-spdifrx.txt | 56 - .../bindings/sound/st,stm32-spdifrx.yaml | 80 + .../devicetree/bindings/sound/tas2562.txt | 2 +- .../devicetree/bindings/sound/tlv320adcx140.yaml | 82 + Documentation/sound/soc/codec-to-codec.rst | 9 +- MAINTAINERS | 15 +- drivers/gpu/drm/mediatek/mtk_hdmi.c | 54 +- drivers/soundwire/qcom.c | 7 +- drivers/soundwire/stream.c | 16 +- drivers/spi/Kconfig | 2 +- include/dt-bindings/sound/meson-aiu.h | 18 + include/dt-bindings/sound/meson-g12a-toacodec.h | 10 + include/linux/soundwire/sdw.h | 2 + include/sound/compress_driver.h | 40 +- include/sound/hdaudio.h | 2 + include/sound/pcm.h | 14 +- include/sound/rt5682.h | 10 + include/sound/soc-acpi.h | 39 +- include/sound/soc-dai.h | 44 +- include/sound/soc-dapm.h | 6 + include/sound/soc-dpcm.h | 20 +- include/sound/soc.h | 44 +- include/sound/sof/dai-intel.h | 18 +- include/sound/sof/header.h | 11 + include/sound/sof/info.h | 22 +- include/sound/sof/topology.h | 3 +- include/uapi/sound/compress_offload.h | 2 +- include/uapi/sound/compress_params.h | 37 +- include/uapi/sound/sof/abi.h | 2 +- sound/arm/pxa2xx-pcm-lib.c | 8 +- sound/core/compress_offload.c | 42 + sound/core/pcm_dmaengine.c | 6 +- sound/core/pcm_misc.c | 18 +- sound/soc/amd/Kconfig | 10 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 376 +++++ sound/soc/amd/raven/acp3x-i2s.c | 44 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/amd/raven/pci-acp3x.c | 7 +- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 8 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 4 + sound/soc/bcm/bcm63xx-i2s-whistler.c | 317 ++++ sound/soc/bcm/bcm63xx-i2s.h | 90 ++ sound/soc/bcm/bcm63xx-pcm-whistler.c | 485 ++++++ sound/soc/bcm/cygnus-pcm.c | 22 +- sound/soc/cirrus/edb93xx.c | 4 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/Kconfig | 627 ++++---- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cros_ec_codec.c | 25 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs47l15.c | 4 +- sound/soc/codecs/cs47l24.c | 6 +- sound/soc/codecs/cs47l35.c | 6 +- sound/soc/codecs/cs47l85.c | 6 +- sound/soc/codecs/cs47l90.c | 6 +- sound/soc/codecs/cs47l92.c | 4 +- sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/codecs/max98357a.c | 37 +- sound/soc/codecs/mt6660.c | 81 +- sound/soc/codecs/rk3328_codec.c | 31 +- sound/soc/codecs/rl6231.c | 1 + sound/soc/codecs/rl6231.h | 2 +- sound/soc/codecs/rt1015.c | 10 +- sound/soc/codecs/rt1308-sdw.c | 38 +- sound/soc/codecs/rt1308-sdw.h | 2 + sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 333 ++++ sound/soc/codecs/rt5682-sdw.h | 20 + sound/soc/codecs/rt5682.c | 1298 ++++++++++++++-- sound/soc/codecs/rt5682.h | 100 +- sound/soc/codecs/tas2562.c | 121 +- sound/soc/codecs/tas2562.h | 12 +- sound/soc/codecs/tlv320adcx140.c | 920 +++++++++++ sound/soc/codecs/tlv320adcx140.h | 131 ++ sound/soc/codecs/wcd9335.c | 18 +- sound/soc/codecs/wcd9335.h | 7 +- sound/soc/codecs/wcd934x.c | 37 +- sound/soc/codecs/wm0010.c | 2 +- sound/soc/codecs/wm5110.c | 6 +- sound/soc/codecs/wm8974.c | 8 - sound/soc/codecs/wm_adsp.c | 14 +- sound/soc/codecs/wsa881x.c | 46 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/fsl_asrc_dma.c | 10 +- sound/soc/fsl/fsl_spdif.c | 10 +- sound/soc/fsl/fsl_ssi.c | 8 +- sound/soc/fsl/imx-audmix.c | 8 +- sound/soc/fsl/imx-mc13783.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 4 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/fsl/p1022_rdk.c | 4 +- sound/soc/fsl/wm1133-ev1.c | 6 +- sound/soc/generic/simple-card-utils.c | 60 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +- sound/soc/intel/atom/sst/sst_pci.c | 2 +- sound/soc/intel/boards/Kconfig | 57 +- sound/soc/intel/boards/Makefile | 12 +- sound/soc/intel/boards/bdw-rt5650.c | 15 +- sound/soc/intel/boards/bdw-rt5677.c | 15 +- sound/soc/intel/boards/broadwell.c | 13 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 10 +- sound/soc/intel/boards/bxt_rt298.c | 10 +- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 +- sound/soc/intel/boards/bytcht_da7213.c | 10 +- sound/soc/intel/boards/bytcht_es8316.c | 8 +- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 8 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 8 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 16 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 13 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 13 +- sound/soc/intel/boards/haswell.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 14 +- sound/soc/intel/boards/kbl_rt5660.c | 6 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 12 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 12 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 4 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 27 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 14 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 +- sound/soc/intel/boards/skl_rt286.c | 8 +- sound/soc/intel/boards/sof_da7219_max98373.c | 83 +- sound/soc/intel/boards/sof_maxim_common.c | 80 + sound/soc/intel/boards/sof_maxim_common.h | 24 + sound/soc/intel/boards/sof_pcm512x.c | 448 ++++++ sound/soc/intel/boards/sof_rt5682.c | 137 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 ++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 ++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 ++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 ++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 ++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 7 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 111 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 103 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 34 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 92 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 +- sound/soc/intel/skylake/bxt-sst.c | 3 - sound/soc/intel/skylake/cnl-sst.c | 35 +- sound/soc/intel/skylake/skl-nhlt.c | 3 +- sound/soc/intel/skylake/skl-pcm.c | 20 +- sound/soc/intel/skylake/skl-sst-dsp.h | 2 + sound/soc/intel/skylake/skl.c | 33 +- sound/soc/jz4740/jz4740-i2s.c | 78 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 +- .../soc/mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 23 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 117 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/meson/Kconfig | 41 + sound/soc/meson/Makefile | 19 + sound/soc/meson/aiu-acodec-ctrl.c | 203 +++ sound/soc/meson/aiu-codec-ctrl.c | 151 ++ sound/soc/meson/aiu-encoder-i2s.c | 365 +++++ sound/soc/meson/aiu-encoder-spdif.c | 209 +++ sound/soc/meson/aiu-fifo-i2s.c | 153 ++ sound/soc/meson/aiu-fifo-spdif.c | 186 +++ sound/soc/meson/aiu-fifo.c | 223 +++ sound/soc/meson/aiu-fifo.h | 50 + sound/soc/meson/aiu.c | 388 +++++ sound/soc/meson/aiu.h | 89 ++ sound/soc/meson/axg-card.c | 414 +---- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/g12a-toacodec.c | 252 +++ sound/soc/meson/g12a-tohdmitx.c | 219 +-- sound/soc/meson/gx-card.c | 141 ++ sound/soc/meson/meson-card-utils.c | 385 +++++ sound/soc/meson/meson-card.h | 55 + sound/soc/meson/meson-codec-glue.c | 149 ++ sound/soc/meson/meson-codec-glue.h | 32 + sound/soc/meson/t9015.c | 333 ++++ sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/Kconfig | 22 + sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 14 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 6 +- sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/apq8016_sbc.c | 9 +- sound/soc/qcom/apq8096.c | 6 +- sound/soc/qcom/lpass-platform.c | 4 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 173 ++- sound/soc/qcom/qdsp6/q6asm.c | 243 ++- sound/soc/qcom/qdsp6/q6asm.h | 51 +- sound/soc/qcom/qdsp6/q6routing.c | 21 +- sound/soc/qcom/sdm845.c | 105 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +- sound/soc/rockchip/rockchip_max98090.c | 6 +- sound/soc/rockchip/rockchip_rt5645.c | 6 +- sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/arndale.c | 10 +- sound/soc/samsung/bells.c | 16 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 +- sound/soc/samsung/littlemill.c | 16 +- sound/soc/samsung/lowland.c | 6 +- sound/soc/samsung/neo1973_wm8753.c | 10 +- sound/soc/samsung/odroid.c | 6 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 4 +- sound/soc/samsung/smdk_wm8994pcm.c | 6 +- sound/soc/samsung/snow.c | 8 +- sound/soc/samsung/spdif.c | 8 +- sound/soc/samsung/speyside.c | 10 +- sound/soc/samsung/tm2_wm5110.c | 19 +- sound/soc/samsung/tobermory.c | 10 +- sound/soc/sh/dma-sh7760.c | 16 +- sound/soc/sh/fsi.c | 5 +- sound/soc/sh/migor.c | 6 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 5 +- sound/soc/soc-core.c | 290 ++-- sound/soc/soc-dai.c | 18 +- sound/soc/soc-dapm.c | 220 +-- sound/soc/soc-generic-dmaengine-pcm.c | 26 +- sound/soc/soc-pcm.c | 1627 +++++++++----------- sound/soc/soc-topology.c | 24 +- sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 1 + sound/soc/sof/compress.c | 146 ++ sound/soc/sof/compress.h | 31 + sound/soc/sof/core.c | 10 +- sound/soc/sof/debug.c | 226 +++ sound/soc/sof/imx/imx8.c | 57 +- sound/soc/sof/intel/Kconfig | 20 +- sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 51 +- sound/soc/sof/intel/hda-codec.c | 11 +- sound/soc/sof/intel/hda-compress.c | 114 ++ sound/soc/sof/intel/hda-ctrl.c | 40 +- sound/soc/sof/intel/hda-dai.c | 130 +- sound/soc/sof/intel/hda-dsp.c | 331 +++- sound/soc/sof/intel/hda-ipc.c | 24 +- sound/soc/sof/intel/hda-loader.c | 40 +- sound/soc/sof/intel/hda-pcm.c | 8 +- sound/soc/sof/intel/hda-stream.c | 27 +- sound/soc/sof/intel/hda.c | 433 +++++- sound/soc/sof/intel/hda.h | 120 +- sound/soc/sof/ipc.c | 41 +- sound/soc/sof/loader.c | 6 - sound/soc/sof/ops.h | 59 +- sound/soc/sof/pcm.c | 19 +- sound/soc/sof/pm.c | 176 +-- sound/soc/sof/probe.c | 290 ++++ sound/soc/sof/probe.h | 85 + sound/soc/sof/sof-audio.c | 59 +- sound/soc/sof/sof-audio.h | 6 +- sound/soc/sof/sof-of-dev.c | 10 + sound/soc/sof/sof-priv.h | 71 +- sound/soc/sof/topology.c | 25 +- sound/soc/sprd/Kconfig | 2 +- sound/soc/sprd/sprd-mcdt.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 4 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 +- sound/soc/stm/stm32_i2s.c | 75 +- sound/soc/stm/stm32_sai.c | 26 +- sound/soc/stm/stm32_sai_sub.c | 13 +- sound/soc/stm/stm32_spdifrx.c | 89 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 3 - sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 24 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/Kconfig | 8 +- sound/soc/ti/Makefile | 2 + sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-mcasp.c | 13 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/ti/udma-pcm.c | 43 + sound/soc/ti/udma-pcm.h | 18 + sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/uniphier/aio-compress.c | 22 +- sound/soc/uniphier/aio-dma.c | 6 +- sound/soc/ux500/mop500_ab8500.c | 6 +- sound/soc/ux500/ux500_pcm.c | 8 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- sound/soc/zte/zx-spdif.c | 1 - sound/soc/zte/zx-tdm.c | 3 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 379 files changed, 17570 insertions(+), 3797 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,aiu.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt delete mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/ingenic,aic.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 include/dt-bindings/sound/meson-aiu.h create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h create mode 100644 sound/soc/amd/acp3x-rt5682-max9836.c create mode 100644 sound/soc/bcm/bcm63xx-i2s-whistler.c create mode 100644 sound/soc/bcm/bcm63xx-i2s.h create mode 100644 sound/soc/bcm/bcm63xx-pcm-whistler.c create mode 100644 sound/soc/codecs/rt5682-sdw.c create mode 100644 sound/soc/codecs/rt5682-sdw.h create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h create mode 100644 sound/soc/intel/boards/sof_maxim_common.c create mode 100644 sound/soc/intel/boards/sof_maxim_common.h create mode 100644 sound/soc/intel/boards/sof_pcm512x.c create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c create mode 100644 sound/soc/meson/aiu-acodec-ctrl.c create mode 100644 sound/soc/meson/aiu-codec-ctrl.c create mode 100644 sound/soc/meson/aiu-encoder-i2s.c create mode 100644 sound/soc/meson/aiu-encoder-spdif.c create mode 100644 sound/soc/meson/aiu-fifo-i2s.c create mode 100644 sound/soc/meson/aiu-fifo-spdif.c create mode 100644 sound/soc/meson/aiu-fifo.c create mode 100644 sound/soc/meson/aiu-fifo.h create mode 100644 sound/soc/meson/aiu.c create mode 100644 sound/soc/meson/aiu.h create mode 100644 sound/soc/meson/g12a-toacodec.c create mode 100644 sound/soc/meson/gx-card.c create mode 100644 sound/soc/meson/meson-card-utils.c create mode 100644 sound/soc/meson/meson-card.h create mode 100644 sound/soc/meson/meson-codec-glue.c create mode 100644 sound/soc/meson/meson-codec-glue.h create mode 100644 sound/soc/meson/t9015.c create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h create mode 100644 sound/soc/ti/udma-pcm.c create mode 100644 sound/soc/ti/udma-pcm.h

2 1

[PATCH 0/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 30 Mar '20

30 Mar '20

Hi, Mark I resend the patch and add recipient Vinod as your comment. Thank you. I'd like to send a patch to fix a problem using soc-compress with DPCM. soc_compr_open_fe() and soc_compr_free_fe() call snd_soc_runtime_activate() and snd_soc_runtime_deactivate() but don't lock card->pcm_mutex. It can cause lockdep error, because snd_soc_runtime_activate/deactivate() checks whether pcm_mutex is held using lockdep_assert_held(). I'd like to send a patch which adds mutex_lock/unlock() before and after of the snd_soc_runtime_activate/deactivate() call. If there is anything should be changed in my patch or email, please let me know it. I'll really appreciate it. Lee Gyeongtaek Lee (1): ASoC: soc-compress: lock pcm_mutex to resolve lockdep error sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 0

[PATCH 1/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 30 Mar '20

30 Mar '20

snd_soc_runtime_activate() and snd_soc_runtime_deactivate() require pcm_mutex lock. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 392a1c5b15d3..42d416ac7e9b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -207,7 +207,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -285,7 +287,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) else stream = SNDRV_PCM_STREAM_CAPTURE; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; -- 2.21.0

2 2

[RFC PATCH v1 13/50] Avoid some useless msecs/jiffies conversions
by George Spelvin 30 Mar '20

30 Mar '20

Rather than generating a random number of milliseconds in a constant range and then converting to jiffies, convert the range to jiffies (evaluated at compile time) and choose a random number of jiffies in that range. Likewise, "msecs_to_jiffies(seconds * 1000)" is more conveniently written "seconds * HZ". Signed-off-by: George Spelvin <lkml(a)sdf.org> Cc: Hannes Reinecke <hare(a)suse.de> Cc: linux-scsi(a)vger.kernel.org Cc: Marek Lindner <mareklindner(a)neomailbox.ch> Cc: Simon Wunderlich <sw(a)simonwunderlich.de> Cc: Antonio Quartulli <a(a)unstable.cc> Cc: Sven Eckelmann <sven(a)narfation.org> Cc: b.a.t.m.a.n(a)lists.open-mesh.org Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-wireless(a)vger.kernel.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- drivers/scsi/fcoe/fcoe_ctlr.c | 10 +++++----- net/batman-adv/bat_iv_ogm.c | 2 +- net/batman-adv/bat_v_ogm.c | 8 ++++---- net/bluetooth/hci_request.c | 2 +- net/wireless/scan.c | 2 +- sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- 7 files changed, 14 insertions(+), 14 deletions(-) diff --git a/drivers/scsi/fcoe/fcoe_ctlr.c b/drivers/scsi/fcoe/fcoe_ctlr.c index 1791a393795da..9c530f8827518 100644 --- a/drivers/scsi/fcoe/fcoe_ctlr.c +++ b/drivers/scsi/fcoe/fcoe_ctlr.c @@ -2238,10 +2238,10 @@ static void fcoe_ctlr_vn_restart(struct fcoe_ctlr *fip) if (fip->probe_tries < FIP_VN_RLIM_COUNT) { fip->probe_tries++; - wait = prandom_u32() % FIP_VN_PROBE_WAIT; + wait = prandom_u32_max(msecs_to_jiffies(FIP_VN_PROBE_WAIT)); } else - wait = FIP_VN_RLIM_INT; - mod_timer(&fip->timer, jiffies + msecs_to_jiffies(wait)); + wait = msecs_to_jiffies(FIP_VN_RLIM_INT); + mod_timer(&fip->timer, jiffies + wait); fcoe_ctlr_set_state(fip, FIP_ST_VNMP_START); } @@ -3132,8 +3132,8 @@ static void fcoe_ctlr_vn_timeout(struct fcoe_ctlr *fip) fcoe_ctlr_vn_send(fip, FIP_SC_VN_BEACON, fcoe_all_vn2vn, 0); fip->port_ka_time = jiffies + - msecs_to_jiffies(FIP_VN_BEACON_INT + - (prandom_u32() % FIP_VN_BEACON_FUZZ)); + msecs_to_jiffies(FIP_VN_BEACON_INT) + + prandom_u32_max(msecs_to_jiffies(FIP_VN_BEACON_FUZZ)); } if (time_before(fip->port_ka_time, next_time)) next_time = fip->port_ka_time; diff --git a/net/batman-adv/bat_iv_ogm.c b/net/batman-adv/bat_iv_ogm.c index 46c5cd9f8019e..9efd96e91d53e 100644 --- a/net/batman-adv/bat_iv_ogm.c +++ b/net/batman-adv/bat_iv_ogm.c @@ -288,7 +288,7 @@ batadv_iv_ogm_emit_send_time(const struct batadv_priv *bat_priv) /* when do we schedule a ogm packet to be sent */ static unsigned long batadv_iv_ogm_fwd_send_time(void) { - return jiffies + msecs_to_jiffies(prandom_u32() % (BATADV_JITTER / 2)); + return jiffies + prandom_u32_max(msecs_to_jiffies(BATADV_JITTER / 2)); } /* apply hop penalty for a normal link */ diff --git a/net/batman-adv/bat_v_ogm.c b/net/batman-adv/bat_v_ogm.c index 411ce5fc6492f..61fa742ff5501 100644 --- a/net/batman-adv/bat_v_ogm.c +++ b/net/batman-adv/bat_v_ogm.c @@ -85,12 +85,12 @@ struct batadv_orig_node *batadv_v_ogm_orig_get(struct batadv_priv *bat_priv, */ static void batadv_v_ogm_start_queue_timer(struct batadv_hard_iface *hard_iface) { - unsigned int msecs = BATADV_MAX_AGGREGATION_MS * 1000; + unsigned int delay = msecs_to_jiffies(BATADV_MAX_AGGREGATION_MS); - /* msecs * [0.9, 1.1] */ - msecs += prandom_u32() % (msecs / 5) - (msecs / 10); + /* delay * [0.9, 1.1] */ + delay += prandom_u32_max(delay / 5) - (delay / 10); queue_delayed_work(batadv_event_workqueue, &hard_iface->bat_v.aggr_wq, - msecs_to_jiffies(msecs / 1000)); + delay); } /** diff --git a/net/bluetooth/hci_request.c b/net/bluetooth/hci_request.c index 2a1b64dbf76e4..8b46e23b4abe7 100644 --- a/net/bluetooth/hci_request.c +++ b/net/bluetooth/hci_request.c @@ -1505,7 +1505,7 @@ int hci_get_random_address(struct hci_dev *hdev, bool require_privacy, bacpy(rand_addr, &hdev->rpa); - to = msecs_to_jiffies(hdev->rpa_timeout * 1000); + to = hdev->rpa_timeout * HZ; if (adv_instance) queue_delayed_work(hdev->workqueue, &adv_instance->rpa_expired_cb, to); diff --git a/net/wireless/scan.c b/net/wireless/scan.c index aef240fdf8df6..b6856cbb68d3b 100644 --- a/net/wireless/scan.c +++ b/net/wireless/scan.c @@ -700,7 +700,7 @@ void cfg80211_bss_age(struct cfg80211_registered_device *rdev, unsigned long age_secs) { struct cfg80211_internal_bss *bss; - unsigned long age_jiffies = msecs_to_jiffies(age_secs * MSEC_PER_SEC); + unsigned long age_jiffies = age_secs * HZ; spin_lock_bh(&rdev->bss_lock); list_for_each_entry(bss, &rdev->bss_list, list) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e0c1f25..8a2bf333200c1 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1839,7 +1839,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time *= HZ; } } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index df40d38f6e290..1ea763f9f956d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1937,7 +1937,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, long t = runtime->period_size * 2 / runtime->rate; tout = max(t, tout); } - tout = msecs_to_jiffies(tout * 1000); + tout *= HZ; } tout = schedule_timeout(tout); -- 2.26.0

3 9

[PATCH 1/1] ASoC: dpcm: allow start or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. To fix this problem, SND_SOC_DPCM_STATE_PAUSED should be allowed also for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca0..8f6f0ad50288 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.21.0

1 0

[PATCH 0/1] ASoC: dpcm: allow started or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

Hi, When compress offload is used with DPCM, error can be occurred on start or stop after pause. soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. Any feedback will be greatly appreciated. Gyeongtaek Lee (1): ASoC: dpcm: allow start or stop during pause for backend sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 0

[PATCH 0/4] ASoC: remove rtd->cpu/codec_dai{s} v2
by Kuninori Morimoto 30 Mar '20

30 Mar '20

Hi Mark These are rebased "remove rtd->cpu/codec_dai" patches. Kuninori Morimoto (4): ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 26 +++++++++-------- sound/soc/soc-compress.c | 36 +++++++++++------------ sound/soc/soc-core.c | 42 ++++++++++----------------- sound/soc/soc-dapm.c | 4 +-- sound/soc/soc-generic-dmaengine-pcm.c | 6 ++-- sound/soc/soc-pcm.c | 30 +++++++++---------- 6 files changed, 68 insertions(+), 76 deletions(-) -- 2.17.1

1 4

[PATCH v2] fix some output pcms being namehint-ed as input pcms
by sylvain.bertrand＠gmail.com 29 Mar '20

29 Mar '20

This is additional cleanup following: commit d434638a683c01bd6decf8f41863044055237a50 namehint: remember the direction from the upper levels signed-off-by: Sylvain BERTRAND <sylvain.bertrand(a)legeek.net> --- based on arecord -L output --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

1 0

[PATCH] ASoC: bcm2835-i2s: substream alignment now independent in hwparams
by Matt Flax 29 Mar '20

29 Mar '20

Substream sample alignment was being set in hwparams for both substreams at the same time. This became a problem when the Audio Injector isolated sound card needed to offset sample alignment for high sample rates. The latency difference between playback and capture occurs because of the digital isolation chip propagation time, particularly when the codec is master and the DAC return is twice delayed. This patch sets sample alignment registers based on the substream direction in hwparams. This gives the machine driver more control over sample alignment in the bcm2835 i2s driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/bcm/bcm2835-i2s.c | 36 +++++++++++++++++++----------------- 1 file changed, 19 insertions(+), 17 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index e6a12e271b07..9db542699a13 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -493,11 +493,6 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, - rx_mask, slot_width, data_delay, odd_slot_offset); - bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, - tx_mask, slot_width, data_delay, odd_slot_offset); - /* * Transmitting data immediately after frame start, eg * in left-justified or DSP mode A, only works stable @@ -508,19 +503,26 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, "Unstable slave config detected, L/R may be swapped"); /* - * Set format for both streams. - * We cannot set another frame length - * (and therefore word length) anyway, - * so the format will be the same. + * Set format on a per stream basis. + * The alignment format can be different depending on direction. */ - regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, - format - | BCM2835_I2S_CH1_POS(rx_ch1_pos) - | BCM2835_I2S_CH2_POS(rx_ch2_pos)); - regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, - format - | BCM2835_I2S_CH1_POS(tx_ch1_pos) - | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, + rx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, + format + | BCM2835_I2S_CH1_POS(rx_ch1_pos) + | BCM2835_I2S_CH2_POS(rx_ch2_pos)); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, + tx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, + format + | BCM2835_I2S_CH1_POS(tx_ch1_pos) + | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + } /* Setup the I2S mode */ -- 2.20.1

3 8

[PATCH v2] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 29 Mar '20

29 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 76f7dec08fd6 (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

2 1

about alc880 no sound and eapd coef
by 尤晓杰 29 Mar '20

29 Mar '20

dear friend, I am using ecs g420 laptop, use options snd-hda-intel model=generic the speaker and mic correctly, 2 pop sounds,when the kernel load(left channel) and the x11 load(right channel), but with standard way only one pop sound when kernel load and silent always. the mic is ok. I have got datasheet of alc880 and see the eapd function, but in your code I see this need coef parameter. so what should I write using verb to open eapd? the attachment is the schematic of sound part.in summary: the 0x14 hp out, 0x18 mic in,0x1c cd in, 0x1e spif out, no jack sense. eapd shared with spdif in. all are correct with default of datasheet's describe. I use verb to dectt each pin widget's capability , take 0x14 as exaple, no eapd capability.To set 0x14's eapd with hda-verb /dev/snd/hwC0D0 0x70c 0x2, failed to read it again alwas 0. and bios also reported 0x16, I think it is bios's bug. I what to use "Early Patching" method to deal with it without changing source code, is it feasible? could you give me some idea? thanks!!!! dmesg: [ 8.009944] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC880: line_outs=2 (0x14/0x16/0x0/0x0/0x0) type:hp [ 8.014023] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.017999] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.021625] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 8.024255] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x1e/0x0 [ 8.026741] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 8.038468] mc: Linux media interface: v0.10 [ 8.067515] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 8.078984] snd_hda_codec_realtek hdaudioC0D0: CD=0x1c [ 8.290306] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/input13

2 1

Port HDA Analyzer to python 3
by andythe_great 29 Mar '20

29 Mar '20

Hello, Is it possible to port HDA Analyzer to python 3? https://www.alsa-project.org/main/index.php/HDA_Analyzer Because openSUSE is retiring python 2. I tried using python-modernize https://python-modernize.readthedocs.io/en/latest/ but many could not get it to run. Thanks. Andy

2 1

[PATCH] ALSA: hda: default enable CA0132 DSP support
by Rouven Czerwinski 29 Mar '20

29 Mar '20

From: Rouven Czerwinski <rouven(a)czerwinskis.de> If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as well. Disabled DSP support leads to a hanging alsa system and no sound output on the card otherwise. Tested on: 06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01) Signed-off-by: Rouven Czerwinski <rouven(a)czerwinskis.de> --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d7..e1d3082a4fe9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help base-commit: e595dd94515ed6bc5ba38fce0f9598db8c0ee9a9 -- 2.25.1

2 1

[alsa-devel] [PATCH 0/3] ALSA: bebob: misc improvements for starting/stopping packet streaming
by Takashi Sakamoto 27 Mar '20

27 Mar '20

Hi, This patchset improves model-specific issues in a point of protocol version supported by the model. The changes are relevant to packet streaming, thus it can brings some regressions. For testers, I prepare for a remote branch to backport the latest codes to Linux v4.17 or later. I'm happy to receive any report. https://github.com/takaswie/snd-firewire-improve/tree/topic/bebob-connectio… Takashi Sakamoto (3): ALSA: bebob: expand sleep just after breaking connections for protocol version 1 ALSA: bebob: expand delay of start for IR context just for version 3 firmware ALSA: bebob: link the order of establishing connections and Syt-match clock mode sound/firewire/bebob/bebob_stream.c | 76 ++++++++++++++--------------- 1 file changed, 38 insertions(+), 38 deletions(-) -- 2.20.1

3 7

[PATCH] ASoC: topology: Add missing memory checks
by Amadeusz Sławiński 27 Mar '20

27 Mar '20

kstrdup is an allocation function and it can fail, so its return value should be checked and handled appropriately. In order to check all cases, we need to modify set_stream_info to return a value, so check that everything went correctly when doing kstrdup(). Later add proper checks and error handlers. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 65 +++++++++++++++++++++++++++++++--------- 1 file changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..0bec3ff782c1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1766,10 +1766,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1780,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1817,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1849,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1857,9 +1867,17 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (ret != 0) { dev_err(dai->dev, "Failed to create DAI widgets %d\n", ret); snd_soc_unregister_dai(dai); - return ret; + goto err; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1934,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2436,13 +2463,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2485,16 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + + return ret; } /* load physical DAI elements */ -- 2.17.1

2 2

[PATCH 0/4] ASoC: Intel: add SoundWire machine driver
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

To handle multiple hardware combinations, this patchset suggests a single machine driver which will create and initialize dailinks dynamically. This allows us to support new configurations easily, as shown with the TigerLake rt5682 example. Each configuration updates the card component string, and UCM can test for the presence of components to configure them as needed. Since we use a single the machine driver name, all previous ACPI tables need to be updated. That should have no impact since the machine drivers listed at the time were not upstreamed and are no longer maintained. Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Pierre-Louis Bossart (1): ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (1): ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms sound/soc/intel/boards/Kconfig | 24 + sound/soc/intel/boards/Makefile | 8 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 +++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 +++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 +++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 +++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 +++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + .../intel/common/soc-acpi-intel-cml-match.c | 24 +- .../intel/common/soc-acpi-intel-icl-match.c | 6 +- .../intel/common/soc-acpi-intel-tgl-match.c | 30 +- 14 files changed, 1896 insertions(+), 11 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c -- 2.20.1

2 11

[PATCH] dt-bindings: sound: tlv320adcx140: Remove undocumented property
by Dan Murphy 27 Mar '20

27 Mar '20

Remove undocumented and unneeded ti,use-internal-reg from the example as it was an artifact from initial development. The code does not query for this property and as the document indicates if areg-supply is undefined then the internal regulator is used. Fixes: 302c0b7490cd ("dt-bindings: sound: Add TLV320ADCx140 dt bindings") CC: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 1433ff62b14f..ab2268c0ee67 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -76,7 +76,6 @@ examples: codec: codec@4c { compatible = "ti,tlv320adc5140"; reg = <0x4c>; - ti,use-internal-areg; ti,mic-bias-source = <6>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; }; -- 2.25.1

2 1

[PATCH] ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
by Colin King 27 Mar '20

27 Mar '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of statements that are not indented correctly, add in the missing tab and break the lines to address a checkpatch warning. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d62c0d90c41e..e362f0bc9e46 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -458,7 +458,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_ITER; frmt_val = mmACP_I2STDM_TXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { struct i2s_stream_instance *rtd = @@ -474,7 +475,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_IRER; frmt_val = mmACP_I2STDM_RXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->tdm_mode == TDM_ENABLE) { rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val); -- 2.25.1

2 1

[PATCH] additional cleanup for "arecord -L show entries which are not capture pcms"
by sylvain.bertrand＠gmail.com 27 Mar '20

27 Mar '20

I did post to github, my bad. On Fri, Mar 27, 2020 at 04:01:57AM -0700, Jaroslav Kysela wrote: > Closed #39 via d434638a683c01bd6decf8f41863044055237a50. Tested with the mic from my usb webcam, and it actually needs a little bit more tidying. -- Sylvain --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

1 0

route and chmap
by Jonas Holmberg 27 Mar '20

27 Mar '20

Hi, I use the route plugin to copy mono to both FL and FR (front left and front right stereo) of the slave during playback. When get_chmap() is called in the route plugin it will call get_chmap() of the slave, which returns 2 channels with FL and FR positions, and then route itself returns 1 channel with FL position. But the application that called snd_pcm_get_chmap() on the playback device expected to get position MONO instead of FL since the number of channels is 1. Is the application wrong in not accepting the chmap or should the route plugin have changed position from FL to MONO? Is it possible to add something to the config (asound.conf) to make FL become MONO in the reply of get_chmap() or solve the problem in any other way? BR /Jonas

2 2

[PATCH 00/36] ASoC: remove rtd->cpu/codec_dai{s}
by Kuninori Morimoto 27 Mar '20

27 Mar '20

Hi Mark Now, CPU/Codec DAI(s) were replaced by rtd->dais. Thus, We don't need rtd->cpu/codec_dai{s} anymore. This pathset replaces it by new macro. Kuninori Morimoto (36): ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 30 +++++++------ sound/arm/pxa2xx-pcm-lib.c | 8 ++-- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +-- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 16 +++---- sound/soc/bcm/cygnus-pcm.c | 22 +++++----- sound/soc/cirrus/edb93xx.c | 4 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/cs47l15.c | 4 +- sound/soc/codecs/cs47l24.c | 6 +-- sound/soc/codecs/cs47l35.c | 6 +-- sound/soc/codecs/cs47l85.c | 6 +-- sound/soc/codecs/cs47l90.c | 6 +-- sound/soc/codecs/cs47l92.c | 4 +- sound/soc/codecs/wm5110.c | 6 +-- sound/soc/codecs/wm_adsp.c | 10 ++--- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 10 ++--- sound/soc/fsl/fsl_asrc_dma.c | 6 +-- sound/soc/fsl/fsl_spdif.c | 10 ++--- sound/soc/fsl/fsl_ssi.c | 8 ++-- sound/soc/fsl/imx-audmix.c | 8 ++-- sound/soc/fsl/imx-mc13783.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 ++--- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 4 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/fsl/p1022_rdk.c | 4 +- sound/soc/fsl/wm1133-ev1.c | 6 +-- sound/soc/generic/simple-card-utils.c | 12 +++--- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +-- sound/soc/intel/boards/bdw-rt5650.c | 6 +-- sound/soc/intel/boards/bdw-rt5677.c | 6 +-- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/bxt_rt298.c | 8 ++-- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 ++--- sound/soc/intel/boards/bytcht_da7213.c | 8 ++-- sound/soc/intel/boards/bytcht_es8316.c | 8 ++-- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 8 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +-- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 ++-- sound/soc/intel/boards/cml_rt1011_rt5682.c | 6 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 10 ++--- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +-- sound/soc/intel/boards/kbl_rt5660.c | 6 +-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 ++-- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 8 ++-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 12 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +++---- sound/soc/intel/boards/skl_rt286.c | 8 ++-- sound/soc/intel/boards/sof_da7219_max98373.c | 8 ++-- sound/soc/intel/boards/sof_pcm512x.c | 8 ++-- sound/soc/intel/boards/sof_rt5682.c | 6 +-- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 ++++++------ sound/soc/intel/skylake/skl-pcm.c | 10 ++--- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 ++--- .../mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 6 +-- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 4 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-card.c | 8 ++-- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 8 ++-- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 6 +-- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/apq8096.c | 6 +-- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 22 +++++----- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +++---- sound/soc/rockchip/rockchip_max98090.c | 6 +-- sound/soc/rockchip/rockchip_rt5645.c | 6 +-- sound/soc/samsung/arndale.c | 6 +-- sound/soc/samsung/bells.c | 16 +++---- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 +- sound/soc/samsung/littlemill.c | 14 +++---- sound/soc/samsung/lowland.c | 4 +- sound/soc/samsung/neo1973_wm8753.c | 10 ++--- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +-- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 +- sound/soc/samsung/snow.c | 4 +- sound/soc/samsung/spdif.c | 8 ++-- sound/soc/samsung/speyside.c | 8 ++-- sound/soc/samsung/tm2_wm5110.c | 16 +++---- sound/soc/samsung/tobermory.c | 8 ++-- sound/soc/sh/dma-sh7760.c | 16 +++---- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 6 +-- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 36 ++++++++-------- sound/soc/soc-core.c | 42 +++++++------------ sound/soc/soc-dapm.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 6 +-- sound/soc/soc-pcm.c | 30 ++++++------- sound/soc/sof/intel/hda-dai.c | 6 +-- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 4 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 +++--- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 6 +-- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +-- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/uniphier/aio-compress.c | 22 +++++----- sound/soc/uniphier/aio-dma.c | 6 +-- sound/soc/ux500/mop500_ab8500.c | 6 +-- sound/soc/ux500/ux500_pcm.c | 8 ++-- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 191 files changed, 573 insertions(+), 577 deletions(-) -- 2.17.1

3 72

[PATCH v2 00/11] ASoC: SOF: Intel: add SoundWire support
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

This patchset provides the support for SoundWire support on Intel CometLake, IcelLake and TigerLake RVP platforms and form-factor devices to be released 'soon'. The bulk of the code is about detecting a valid SoundWire configuration from ACPI, and implementing the interfaces suggested in '[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces' for interrupts, PCI wakes and clock-stop configurations. Since that SoundWire series will not be in 5.7, the build support for SOF w/ SoundWire is not provided for now, and fall-back functions will be used. This code is tested on a daily basis in the SOF tree and is not expected to change in significant ways. Changes since v2: Corrected error in ACPI table (thanks Amadeusz) Added patch 11 to add reset cycle required on some SoundWire platforms Bard Liao (1): ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Pierre-Louis Bossart (8): ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Rander Wang (2): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread include/sound/soc-acpi.h | 39 +- include/sound/sof/dai-intel.h | 18 +- .../intel/common/soc-acpi-intel-cml-match.c | 87 +++- .../intel/common/soc-acpi-intel-icl-match.c | 97 ++++- .../intel/common/soc-acpi-intel-tgl-match.c | 49 ++- sound/soc/sof/intel/hda-ctrl.c | 25 +- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 31 ++ sound/soc/sof/intel/hda.c | 400 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 66 +++ 10 files changed, 750 insertions(+), 64 deletions(-) -- 2.20.1

2 14

Applied "ASoC: SOF: Intel: add SoundWire configuration interface" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: add SoundWire configuration interface has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 51dfed1e178a38202960b98f6e29df009a06050f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:18 -0500 Subject: [PATCH] ASoC: SOF: Intel: add SoundWire configuration interface Now that the SoundWire core supports the multi-step initialization, call the relevant APIs. The actual hardware enablement can be done in two places, ideally we'd want to startup the SoundWire IP as soon as possible (while still taking power rail dependencies into account) However when suspend/resume is implemented, the DSP device will be resumed first, and only when the DSP firmware is downloaded/booted would the SoundWire child devices be resumed, so there are only marginal benefits in starting the IP earlier for the first probe. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-loader.c | 13 ++++ sound/soc/sof/intel/hda.c | 120 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 44 ++++++++++++ 3 files changed, 177 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 0633b76dab49..2ae94ea53122 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -399,6 +399,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) /* post fw run operations */ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) { + int ret; + + if (sdev->first_boot) { + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: could not startup SoundWire links\n"); + return ret; + } + } + + hda_sdw_int_enable(sdev, true); + /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee1edb53840c..c1fe94800da1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -18,7 +18,9 @@ #include <sound/hdaudio_ext.h> #include <sound/hda_register.h> +#include <linux/acpi.h> #include <linux/module.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> @@ -34,6 +36,98 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); +} + +static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(sdev->dev); + + /* save ACPI info for the probe step */ + hdev = sdev->pdata->hw_pdata; + + ret = sdw_intel_acpi_scan(handle, &hdev->info); + if (ret < 0) { + dev_err(sdev->dev, "%s failed\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_res res; + acpi_handle handle; + void *sdw; + + handle = ACPI_HANDLE(sdev->dev); + + hdev = sdev->pdata->hw_pdata; + + memset(&res, 0, sizeof(res)); + + res.mmio_base = sdev->bar[HDA_DSP_BAR]; + res.irq = sdev->ipc_irq; + res.handle = hdev->info.handle; + res.parent = sdev->dev; + + /* + * ops and arg fields are not populated for now, + * they will be needed when the DAI callbacks are + * provided + */ + + /* we could filter links here if needed, e.g for quirks */ + res.count = hdev->info.count; + res.link_mask = hdev->info.link_mask; + + sdw = sdw_intel_probe(&res); + if (!sdw) { + dev_err(sdev->dev, "error: SoundWire probe failed\n"); + return -EINVAL; + } + + /* save context */ + hdev->sdw = sdw; + + return 0; +} + +int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + return sdw_intel_startup(hdev->sdw); +} + +static int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + hda_sdw_int_enable(sdev, false); + + if (hdev->sdw) + sdw_intel_exit(hdev->sdw); + hdev->sdw = NULL; + + return 0; +} +#endif + /* * Debug */ @@ -346,9 +440,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, static int hda_init_caps(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_sof_pdata *pdata = sdev->pdata; #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_ext_link *hlink; #endif + struct sof_intel_hda_dev *hdev = pdata->hw_pdata; + u32 link_mask; int ret = 0; device_disable_async_suspend(bus->dev); @@ -365,6 +462,27 @@ static int hda_init_caps(struct snd_sof_dev *sdev) return ret; } + /* scan SoundWire capabilities exposed by DSDT */ + ret = hda_sdw_acpi_scan(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire ACPI scan error\n"); + return ret; + } + + link_mask = hdev->info.link_mask; + if (!link_mask) { + /* + * probe/allocated SoundWire resources. + * The hardware configuration takes place in hda_sdw_startup + * after power rails are enabled. + */ + ret = hda_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + return ret; + } + } + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -622,6 +740,8 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) snd_hdac_ext_bus_device_remove(bus); #endif + hda_sdw_exit(sdev); + if (!IS_ERR_OR_NULL(hda->dmic_dev)) platform_device_unregister(hda->dmic_dev); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 2a57fc9f3e58..928a3432e9e6 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,8 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/compress_driver.h> #include <sound/hda_codec.h> #include <sound/hdaudio_ext.h> @@ -436,6 +438,12 @@ struct sof_intel_hda_dev { /* delayed work to enter D0I3 opportunistically */ struct delayed_work d0i3_work; + + /* ACPI information stored between scan and probe steps */ + struct sdw_intel_acpi_info info; + + /* sdw context allocated by SoundWire driver */ + struct sdw_intel_ctx *sdw; }; static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) @@ -654,6 +662,42 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag); int hda_dsp_trace_release(struct snd_sof_dev *sdev); int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); +/* + * SoundWire support + */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +int hda_sdw_startup(struct snd_sof_dev *sdev); +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); + +#else + +static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ +} + +#endif + /* common dai driver */ extern struct snd_soc_dai_driver skl_dai[]; -- 2.20.1

1 0

Applied "ASoC: SOF: IPC: dai-intel: move ALH declarations in header file" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: IPC: dai-intel: move ALH declarations in header file has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f8e25018801548314b3d908315cbf271e8eceba8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:19 -0500 Subject: [PATCH] ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ALH was inserted in the wrong place during integration, add after DMIC to mirror the file used by SOF firmware. No functional change, just text move in the same file to better track changes, if any. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/sof/dai-intel.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h index 5f1ef5565be6..04e48227f542 100644 --- a/include/sound/sof/dai-intel.h +++ b/include/sound/sof/dai-intel.h @@ -87,6 +87,15 @@ struct sof_ipc_dai_hda_params { uint32_t link_dma_ch; } __packed; +/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ +struct sof_ipc_dai_alh_params { + struct sof_ipc_hdr hdr; + uint32_t stream_id; + + /* reserved for future use */ + uint32_t reserved[15]; +} __packed; + /* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */ /* This struct is defined per 2ch PDM controller available in the platform. @@ -179,13 +188,4 @@ struct sof_ipc_dai_dmic_params { struct sof_ipc_dai_dmic_pdm_ctrl pdm[0]; } __packed; -/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ -struct sof_ipc_dai_alh_params { - struct sof_ipc_hdr hdr; - uint32_t stream_id; - - /* reserved for future use */ - uint32_t reserved[15]; -} __packed; - #endif -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d2c383aa49dce9c5b59930c999099ecc580857fe Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:20 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks These callbacks are invoked when a matching hw_params/hw_free() DAI operation takes place, and will result in IPC operations with the SOF firmware. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 71 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c1fe94800da1..1f93124a63cd 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -24,6 +24,7 @@ #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" @@ -38,6 +39,74 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +static int sdw_params_stream(struct device *dev, + struct sdw_intel_stream_params_data *params_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = params_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = params_data->link_id; + int alh_stream_id = params_data->alh_stream_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | (d->id); + config.alh.stream_id = alh_stream_id; + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n", + link_id, d->id, alh_stream_id); + } + + return ret; +} + +static int sdw_free_stream(struct device *dev, + struct sdw_intel_stream_free_data *free_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = free_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = free_data->link_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | d->id; + config.alh.stream_id = 0xFFFF; /* invalid value on purpose */ + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to free stream for link %d dai->id %d\n", + link_id, d->id); + } + + return ret; +} + +static const struct sdw_intel_ops sdw_callback = { + .params_stream = sdw_params_stream, + .free_stream = sdw_free_stream, +}; + void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); @@ -80,6 +149,8 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.irq = sdev->ipc_irq; res.handle = hdev->info.handle; res.parent = sdev->dev; + res.ops = &sdw_callback; + res.dev = sdev->dev; /* * ops and arg fields are not populated for now, -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3eadff5639b01c17f5f2ffeb209d05cc19706687 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:22 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend Doing this avoid conflicts and errors reported on the bus. The interrupts are only re-enabled on resume after the firmware is downloaded, so the behavior is not fully symmetric Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c396b7ef0328..1aff90042694 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -594,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #endif int ret; + hda_sdw_int_enable(sdev, false); + /* disable IPC interrupts */ hda_dsp_ipc_int_disable(sdev); -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 722ba5f1f530a919d28a0dd9e8e0ec63af18270d Mon Sep 17 00:00:00 2001 From: Bard Liao <yung-chuan.liao(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:23 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers We have a single irq handler for SOF interrupts. We can further merge SoundWire ones to completely remove MSI interrupts handling issues leading to timeouts. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++++++++++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index dea3c385b664..ee4f1ceca883 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -198,6 +198,38 @@ static int hda_sdw_exit(struct snd_sof_dev *sdev) return 0; } + +static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + bool ret = false; + u32 irq_status; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return ret; + + /* store status */ + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2); + + /* invalid message ? */ + if (irq_status == 0xffffffff) + goto out; + + /* SDW message ? */ + if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW) + ret = true; + +out: + return ret; +} + +static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return sdw_intel_thread(irq, context); +} + #endif /* @@ -618,6 +650,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; /* deal with streams and controller first */ if (hda_dsp_check_stream_irq(sdev)) @@ -626,6 +659,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_ipc_irq(sdev)) sof_ops(sdev)->irq_thread(irq, sdev); + if (hda_dsp_check_sdw_irq(sdev)) + hda_dsp_sdw_thread(irq, hdev->sdw); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 928a3432e9e6..fc104c5ba006 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -232,6 +232,8 @@ #define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10) #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) +#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) + /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 #define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00) @@ -696,6 +698,15 @@ static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { } +static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + return false; +} + +static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return IRQ_HANDLED; +} #endif /* common dai driver */ -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 02df8f4364b070428af0e5b6c8739c884c8ad4e7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:24 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks Add module parameter so that the different modes can be quickly tested. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-9-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee4f1ceca883..1e69cfcee8e0 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -40,6 +40,16 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +/* + * The default for SoundWire clock stop quirks is to power gate the IP + * and do a Bus Reset, this will need to be modified when the DSP + * needs to remain in D0i3 so that the Master does not lose context + * and enumeration is not required on clock restart + */ +static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET; +module_param(sdw_clock_stop_quirks, int, 0444); +MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks"); + static int sdw_params_stream(struct device *dev, struct sdw_intel_stream_params_data *params_data) { @@ -149,6 +159,7 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.parent = sdev->dev; res.ops = &sdw_callback; res.dev = sdev->dev; + res.clock_stop_quirks = sdw_clock_stop_quirks; /* * ops and arg fields are not populated for now, -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bbd19cdca8279cf244a301c6a13ae5ec9e4ef976 Mon Sep 17 00:00:00 2001 From: Rander Wang <rander.wang(a)intel.com> Date: Wed, 25 Mar 2020 16:50:25 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire When a SoundWire link is in clock stop state, a Slave device may wake up the Master for some events such as jack detection. The WAKEEN interrupt will be triggered and processed by the audio pci device. If audio device is in D3, the interrupt will be routed to PME, or aggregated at cAVS level as interrupt when audio device is in D0. This patch only supports D3 case, where the audio pci device will be resumed by a PME event and the WAKEEN interrupt will be processed after audio pci device is powered up and ROM is initialized successfully. The WAKEEN handling is only enabled after the first boot due to dependencies on a shim_lock mutex being initialized. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-loader.c | 18 ++++++++++++++++++ sound/soc/sof/intel/hda.c | 11 +++++++++++ sound/soc/sof/intel/hda.h | 5 +++++ 3 files changed, 34 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 2ae94ea53122..e1550ccd0a49 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -346,6 +346,24 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) goto cleanup; } + /* + * When a SoundWire link is in clock stop state, a Slave + * device may trigger in-band wakes for events such as jack + * insertion or acoustic event detection. This event will lead + * to a WAKEEN interrupt, handled by the PCI device and routed + * to PME if the PCI device is in D3. The resume function in + * audio PCI driver will be invoked by ACPI for PME event and + * initialize the device and process WAKEEN interrupt. + * + * The WAKEEN interrupt should be processed ASAP to prevent an + * interrupt flood, otherwise other interrupts, such IPC, + * cannot work normally. The WAKEEN is handled after the ROM + * is initialized successfully, which ensures power rails are + * enabled before accessing the SoundWire SHIM registers + */ + if (!sdev->first_boot) + hda_sdw_process_wakeen(sdev); + /* * at this point DSP ROM has been initialized and * should be ready for code loading and firmware boot diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1e69cfcee8e0..7d1aa4c7d82c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -241,6 +241,17 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return sdw_intel_thread(irq, context); } +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (!hdev->sdw) + return; + + sdw_intel_process_wakeen_event(hdev->sdw); +} + #endif /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index fc104c5ba006..6f1765b1ed1d 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -671,6 +671,7 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); int hda_sdw_startup(struct snd_sof_dev *sdev); void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev); #else @@ -707,6 +708,10 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) { return IRQ_HANDLED; } + +static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ +} #endif /* common dai driver */ -- 2.20.1

1 0

Applied "Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 90de3281c86ae5378e951e84c76c4759390ff34d Mon Sep 17 00:00:00 2001 From: Rander Wang <rander.wang(a)intel.com> Date: Wed, 25 Mar 2020 16:50:26 -0500 Subject: [PATCH] Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread If pci device is in D0, wakeen interrupt will be aggregated at cAVS level as interrupt. This commit check the wakeen status and process it in irq thread Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-11-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 16 ++++++++++++++++ sound/soc/sof/intel/hda.h | 6 ++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7d1aa4c7d82c..211e91e79eae 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -241,6 +241,19 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return sdw_intel_thread(irq, context); } +static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (hdev->sdw && + snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_SNDW_WAKE_STS)) + return true; + + return false; +} + void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; @@ -684,6 +697,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_sdw_irq(sdev)) hda_dsp_sdw_thread(irq, hdev->sdw); + if (hda_sdw_check_wakeen_irq(sdev)) + hda_sdw_process_wakeen(sdev); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 6f1765b1ed1d..e9825798de77 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -233,6 +233,7 @@ #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) #define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) +#define HDA_DSP_REG_SNDW_WAKE_STS 0x2C192 /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 @@ -709,6 +710,11 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return IRQ_HANDLED; } +static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) +{ + return false; +} + static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) { } -- 2.20.1

1 0

[PATCH 0/1] ASoC: convert to use new I2C API
by Wolfram Sang 27 Mar '20

27 Mar '20

We are deprecating calls which return NULL in favor of new variants which return an ERR_PTR. Only build tested. Wolfram Sang (1): ASoC: pxa: magician: convert to use i2c_new_client_device() sound/soc/pxa/magician.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) -- 2.20.1

2 2

[PATCH] ASoC: rt5682: move DAI clock registry to I2S mode
by shumingf＠realtek.com 27 Mar '20

27 Mar '20

From: Shuming Fan <shumingf(a)realtek.com> The SoundWire mode doesn't need the DAI clocks. Therefore, the DAI clock registry moves to I2S mode case. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 39 +++++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d5a9e8a48607..c9268a230daa 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2856,26 +2856,6 @@ static int rt5682_probe(struct snd_soc_component *component) #endif rt5682->component = component; -#ifdef CONFIG_COMMON_CLK - /* Check if MCLK provided */ - rt5682->mclk = devm_clk_get(component->dev, "mclk"); - if (IS_ERR(rt5682->mclk)) { - if (PTR_ERR(rt5682->mclk) != -ENOENT) { - ret = PTR_ERR(rt5682->mclk); - return ret; - } - rt5682->mclk = NULL; - } - - /* Register CCF DAI clock control */ - ret = rt5682_register_dai_clks(component); - if (ret) - return ret; - - /* Initial setup for CCF */ - rt5682->lrck[RT5682_AIF1] = CLK_48; -#endif - if (rt5682->is_sdw) { slave = rt5682->slave; time = wait_for_completion_timeout( @@ -2885,6 +2865,25 @@ static int rt5682_probe(struct snd_soc_component *component) dev_err(&slave->dev, "Initialization not complete, timed out\n"); return -ETIMEDOUT; } + } else { +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } else { + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; + } + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = CLK_48; +#endif } return 0; -- 2.25.1

2 1

[PATCH 0/1] ALSA: ppc: convert to use new I2C API
by Wolfram Sang 27 Mar '20

27 Mar '20

We are deprecating calls which return NULL in favor of new variants which return an ERR_PTR. Only build tested. Wolfram Sang (1): ALSA: ppc: keywest: convert to use i2c_new_client_device() sound/ppc/keywest.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) -- 2.20.1

2 2

[PATCH] ALSA: hda/realtek: Enable mute LED on an HP system
by Kai-Heng Feng 27 Mar '20

27 Mar '20

The system in question uses ALC285, and it uses GPIO 0x04 to control its mute LED. The mic mute LED can be controlled by GPIO 0x01, however the system uses DMIC so we should use that to control mic mute LED. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.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 63e1a56f705b..1ad8c2e2d1af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4008,6 +4008,12 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10); } +static void alc285_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0x04, 0x00); +} + static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5923,6 +5929,7 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC285_FIXUP_HP_GPIO_LED, }; static const struct hda_fixup alc269_fixups[] = { @@ -7061,6 +7068,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC285_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_gpio_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7208,6 +7219,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.17.1

2 1

[hints] arecord -L show entries which are not capture pcms
by GitHub issues - edited 27 Mar '20

27 Mar '20

alsa-project/alsa-lib issue #39 was edited from sylware: arecord -L will show pcms which are not capture pcms (for instance front speakers, surroundX). Don't know if it is fix-able. Wait for topology/ucm port of classic drivers? Issue URL : https://github.com/alsa-project/alsa-lib/issues/39 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 4.4 1/4] ALSA: hda/realtek: Fix pop noise on ALC225
by Sasha Levin 27 Mar '20

27 Mar '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757 ] Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200311061328.17614-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 55bae9e6de27d..76cf438aa339c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6333,6 +6333,8 @@ static int patch_alc269(struct hda_codec *codec) alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.20.1

1 2

[PATCH AUTOSEL 4.9 1/7] ALSA: hda/realtek: Fix pop noise on ALC225
by Sasha Levin 27 Mar '20

27 Mar '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757 ] Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200311061328.17614-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a64612db1f155..d7b33c9c9f40b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6410,6 +6410,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: spec->codec_variant = ALC269_TYPE_ALC225; break; -- 2.20.1

1 3

[PATCH AUTOSEL 4.14 10/10] ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5461e0530c222129dfc941058be114b5cbc00837 ] The return value checks in snd_pcm_plug_alloc() are covered with snd_BUG_ON() macro that may trigger a kernel WARNING depending on the kconfig. But since the error condition can be triggered by a weird user space parameter passed to OSS layer, we shouldn't give the kernel stack trace just for that. As it's a normal error condition, let's remove snd_BUG_ON() macro usage there. Reported-by: syzbot+2a59ee7a9831b264f45e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200312155730.7520-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 6583eb411f82a..40d2d39151bfb 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames); -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 09/10] ALSA: line6: Fix endless MIDI read loop
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit d683469b3c93d7e2afd39e6e1970f24700eb7a68 ] The MIDI input event parser of the LINE6 driver may enter into an endless loop when the unexpected data sequence is given, as it tries to continue the secondary bytes without termination. Also, when the input data is too short, the parser returns a negative error, while the caller doesn't handle it properly. This would lead to the unexpected behavior as well. This patch addresses those issues by checking the return value correctly and handling the one-byte event in the parser properly. The bug was reported by syzkaller. Reported-by: syzbot+cce32521ee0a824c21f7(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com Link: https://lore.kernel.org/r/20200309095922.30269-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/line6/driver.c | 2 +- sound/usb/line6/midibuf.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b223de3defc4f..bf4eacc53a7d2 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -313,7 +313,7 @@ static void line6_data_received(struct urb *urb) line6_midibuf_read(mb, line6->buffer_message, LINE6_MIDI_MESSAGE_MAXLEN); - if (done == 0) + if (done <= 0) break; line6->message_length = done; diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c index 36a610ba342ec..c931d48801ebe 100644 --- a/sound/usb/line6/midibuf.c +++ b/sound/usb/line6/midibuf.c @@ -163,7 +163,7 @@ int line6_midibuf_read(struct midi_buffer *this, unsigned char *data, int midi_length_prev = midibuf_message_length(this->command_prev); - if (midi_length_prev > 0) { + if (midi_length_prev > 1) { midi_length = midi_length_prev - 1; repeat = 1; } else -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 08/10] ALSA: pcm: oss: Avoid plugin buffer overflow
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f2ecf903ef06eb1bbbfa969db9889643d487e73a ] Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Link: https://lore.kernel.org/r/20200309082148.19855-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index b8ab46b8298de..6583eb411f82a 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -209,6 +209,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); @@ -220,6 +222,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 02/10] ALSA: hda/realtek: Fix pop noise on ALC225
by Sasha Levin 27 Mar '20

27 Mar '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757 ] Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200311061328.17614-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5a7afbeb612d5..e5d4765a182b5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7265,6 +7265,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: spec->codec_variant = ALC269_TYPE_ALC225; spec->gen.mixer_nid = 0; /* no loopback on ALC225 ALC295 */ -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 14/15] ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5461e0530c222129dfc941058be114b5cbc00837 ] The return value checks in snd_pcm_plug_alloc() are covered with snd_BUG_ON() macro that may trigger a kernel WARNING depending on the kconfig. But since the error condition can be triggered by a weird user space parameter passed to OSS layer, we shouldn't give the kernel stack trace just for that. As it's a normal error condition, let's remove snd_BUG_ON() macro usage there. Reported-by: syzbot+2a59ee7a9831b264f45e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200312155730.7520-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 9b588c6a6f099..732bbede7ebfd 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames); -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 13/15] ALSA: line6: Fix endless MIDI read loop
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit d683469b3c93d7e2afd39e6e1970f24700eb7a68 ] The MIDI input event parser of the LINE6 driver may enter into an endless loop when the unexpected data sequence is given, as it tries to continue the secondary bytes without termination. Also, when the input data is too short, the parser returns a negative error, while the caller doesn't handle it properly. This would lead to the unexpected behavior as well. This patch addresses those issues by checking the return value correctly and handling the one-byte event in the parser properly. The bug was reported by syzkaller. Reported-by: syzbot+cce32521ee0a824c21f7(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com Link: https://lore.kernel.org/r/20200309095922.30269-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/line6/driver.c | 2 +- sound/usb/line6/midibuf.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index bbcb0d4d83aec..0193d4989107b 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -320,7 +320,7 @@ static void line6_data_received(struct urb *urb) line6_midibuf_read(mb, line6->buffer_message, LINE6_MIDI_MESSAGE_MAXLEN); - if (done == 0) + if (done <= 0) break; line6->message_length = done; diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c index 36a610ba342ec..c931d48801ebe 100644 --- a/sound/usb/line6/midibuf.c +++ b/sound/usb/line6/midibuf.c @@ -163,7 +163,7 @@ int line6_midibuf_read(struct midi_buffer *this, unsigned char *data, int midi_length_prev = midibuf_message_length(this->command_prev); - if (midi_length_prev > 0) { + if (midi_length_prev > 1) { midi_length = midi_length_prev - 1; repeat = 1; } else -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 12/15] ALSA: pcm: oss: Avoid plugin buffer overflow
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f2ecf903ef06eb1bbbfa969db9889643d487e73a ] Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Link: https://lore.kernel.org/r/20200309082148.19855-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 31cb2acf8afcc..9b588c6a6f099 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -209,6 +209,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); @@ -220,6 +222,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 03/15] ALSA: hda/realtek: Fix pop noise on ALC225
by Sasha Levin 27 Mar '20

27 Mar '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757 ] Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200311061328.17614-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 86ab022f386ec..23aab2fdac468 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7828,6 +7828,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 17/19] ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5461e0530c222129dfc941058be114b5cbc00837 ] The return value checks in snd_pcm_plug_alloc() are covered with snd_BUG_ON() macro that may trigger a kernel WARNING depending on the kconfig. But since the error condition can be triggered by a weird user space parameter passed to OSS layer, we shouldn't give the kernel stack trace just for that. As it's a normal error condition, let's remove snd_BUG_ON() macro usage there. Reported-by: syzbot+2a59ee7a9831b264f45e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200312155730.7520-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 9b588c6a6f099..732bbede7ebfd 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames); -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 16/19] ALSA: line6: Fix endless MIDI read loop
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit d683469b3c93d7e2afd39e6e1970f24700eb7a68 ] The MIDI input event parser of the LINE6 driver may enter into an endless loop when the unexpected data sequence is given, as it tries to continue the secondary bytes without termination. Also, when the input data is too short, the parser returns a negative error, while the caller doesn't handle it properly. This would lead to the unexpected behavior as well. This patch addresses those issues by checking the return value correctly and handling the one-byte event in the parser properly. The bug was reported by syzkaller. Reported-by: syzbot+cce32521ee0a824c21f7(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com Link: https://lore.kernel.org/r/20200309095922.30269-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/line6/driver.c | 2 +- sound/usb/line6/midibuf.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b5a3f754a4f13..4f096685ed654 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -305,7 +305,7 @@ static void line6_data_received(struct urb *urb) line6_midibuf_read(mb, line6->buffer_message, LINE6_MIDI_MESSAGE_MAXLEN); - if (done == 0) + if (done <= 0) break; line6->message_length = done; diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c index 8d6eefa0d9362..6a70463f82c4e 100644 --- a/sound/usb/line6/midibuf.c +++ b/sound/usb/line6/midibuf.c @@ -159,7 +159,7 @@ int line6_midibuf_read(struct midi_buffer *this, unsigned char *data, int midi_length_prev = midibuf_message_length(this->command_prev); - if (midi_length_prev > 0) { + if (midi_length_prev > 1) { midi_length = midi_length_prev - 1; repeat = 1; } else -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 15/19] ALSA: pcm: oss: Avoid plugin buffer overflow
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f2ecf903ef06eb1bbbfa969db9889643d487e73a ] Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Link: https://lore.kernel.org/r/20200309082148.19855-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 31cb2acf8afcc..9b588c6a6f099 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -209,6 +209,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); @@ -220,6 +222,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 03/19] ALSA: hda/realtek: Fix pop noise on ALC225
by Sasha Levin 27 Mar '20

27 Mar '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757 ] Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200311061328.17614-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12858d95c2c85..73b182db7ef3b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8080,6 +8080,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.20.1

1 0