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

Re: [PATCH v2 08/21] ASoC: codecs: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 29 Jun '20

29 Jun '20

Hi Peter, Mark Thank you for your review > > static const struct snd_soc_dai_ops tfa9879_dai_ops = { > > .hw_params = tfa9879_hw_params, > > - .digital_mute = tfa9879_digital_mute, > > + .mute_stream = tfa9879_mute, > > .set_fmt = tfa9879_set_fmt, > > + .no_capture_mute = 1, > > }; (snip) > - .digital_mute = tfa9879_digital_mute, > + .mute_stream = tfa9879_mute_stream, Yes, indeed. I can fix it by additional patch if v2 was accepted, or, in v3 if needed. Is it OK for you, and Mark ? Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

[PATCH 10/21] ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 29 Jun '20

29 Jun '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_digital_mute() is internally using both mute_stream() (1) or digital_mute() (2), but the difference between these 2 are only handling direction. We can merge digital_mute() into mute_stream int snd_soc_dai_digital_mute(xxx, int direction) { ... else if (dai->driver->ops->mute_stream) (1) return dai->driver->ops->mute_stream(xxx, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && dai->driver->ops->digital_mute) (2) return dai->driver->ops->digital_mute(xxx); ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This was missed on v2 patch-set. sound/soc/codecs/tas2552.c | 5 +++-- sound/soc/codecs/tas2562.c | 5 +++-- sound/soc/codecs/tas2770.c | 5 +++-- sound/soc/codecs/tas571x.c | 5 +++-- sound/soc/codecs/tas5720.c | 5 +++-- sound/soc/codecs/tas6424.c | 5 +++-- 6 files changed, 18 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 529c0fb93f9b..3a153526e47f 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -465,7 +465,7 @@ static int tas2552_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas2552_mute(struct snd_soc_dai *dai, int mute) +static int tas2552_mute(struct snd_soc_dai *dai, int mute, int direction) { u8 cfg1_reg = 0; struct snd_soc_component *component = dai->component; @@ -519,7 +519,8 @@ static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .set_sysclk = tas2552_set_dai_sysclk, .set_fmt = tas2552_set_dai_fmt, .set_tdm_slot = tas2552_set_dai_tdm_slot, - .digital_mute = tas2552_mute, + .mute_stream = tas2552_mute, + .no_capture_mute = 1, }; /* Formats supported by TAS2552 driver. */ diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 7fae88655a0f..8721b0ecc403 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -334,7 +334,7 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int tas2562_mute(struct snd_soc_dai *dai, int mute) +static int tas2562_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; @@ -552,7 +552,8 @@ static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { .hw_params = tas2562_hw_params, .set_fmt = tas2562_set_dai_fmt, .set_tdm_slot = tas2562_set_dai_tdm_slot, - .digital_mute = tas2562_mute, + .mute_stream = tas2562_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas2562_dai[] = { diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..4538b2d0216f 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -189,7 +189,7 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = { {"VSENSE", "Switch", "VMON"}, }; -static int tas2770_mute(struct snd_soc_dai *dai, int mute) +static int tas2770_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; @@ -530,10 +530,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, } static struct snd_soc_dai_ops tas2770_dai_ops = { - .digital_mute = tas2770_mute, + .mute_stream = tas2770_mute, .hw_params = tas2770_hw_params, .set_fmt = tas2770_set_fmt, .set_tdm_slot = tas2770_set_dai_tdm_slot, + .no_capture_mute = 1, }; #define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 5b7f9fcf6cbf..835a723ce5bc 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -301,7 +301,7 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream, TAS571X_SDI_FMT_MASK, val); } -static int tas571x_mute(struct snd_soc_dai *dai, int mute) +static int tas571x_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u8 sysctl2; @@ -354,7 +354,8 @@ static int tas571x_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops tas571x_dai_ops = { .set_fmt = tas571x_set_dai_fmt, .hw_params = tas571x_hw_params, - .digital_mute = tas571x_mute, + .mute_stream = tas571x_mute, + .no_capture_mute = 1, }; diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index e159f839d928..139ac5e683bf 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -199,7 +199,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, return ret; } -static int tas5720_mute(struct snd_soc_dai *dai, int mute) +static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; @@ -604,7 +604,8 @@ static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = { .hw_params = tas5720_hw_params, .set_fmt = tas5720_set_dai_fmt, .set_tdm_slot = tas5720_set_dai_tdm_slot, - .digital_mute = tas5720_mute, + .mute_stream = tas5720_mute, + .no_capture_mute = 1, }; /* diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aaba39295079..6198138e693a 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -252,7 +252,7 @@ static int tas6424_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas6424_mute(struct snd_soc_dai *dai, int mute) +static int tas6424_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component); @@ -382,7 +382,8 @@ static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = { .hw_params = tas6424_hw_params, .set_fmt = tas6424_set_dai_fmt, .set_tdm_slot = tas6424_set_dai_tdm_slot, - .digital_mute = tas6424_mute, + .mute_stream = tas6424_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas6424_dai[] = { -- 2.25.1

1 0

[PATCH v2 0/7] ASoC: QCOM: Add support for SC7180 lpass variant
by Ajit Pandey 29 Jun '20

29 Jun '20

We've common ASoC cpu driver for QCOM LPASS soc varaints. This patch chain add support for new variant SC7180 soc by doing the required modification in existing common lpass-cpu driver. Below is a brief summary of patch series: PATCH v2 0001 ... 0004: Update lpass-cpu driver and documentation to make it more generic and support newer soc registers configuration. PATCH v2 0005 ... 0007: Add documentation and platform driver for newer SC7180 SOC variant. Ajit Pandey (7): Documentation: device-tree: sound: Update lpass-cpu driver binding ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-cpu: Make "ahbix-clk" an optional clock. ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers include: dt-bindings: sound: Add sc7180-lpass bindings header device-tree: bindings: sound: lpass-cpu: Add new compatible soc ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 ------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 154 +++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 100 ++++++-- sound/soc/qcom/lpass-cpu.c | 116 ++++++---- sound/soc/qcom/lpass-lpaif-reg.h | 203 ++++++++++------- sound/soc/qcom/lpass-platform.c | 86 ++++--- sound/soc/qcom/lpass-sc7180.c | 252 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 40 +++- 11 files changed, 790 insertions(+), 257 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- ('The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,\na Linux Foundation Collaborative Project')

4 18

[PATCH alsa-lib 1/5] Fix invalid "Regex" Type in various Condition blocks
by Hans de Goede 28 Jun '20

28 Jun '20

Regex is not a valid Condition type, using it leads to errors like these: ALSA lib ucm_cond.c:300:(if_eval) unknown If.Condition.Type ALSA lib main.c:983:(snd_use_case_mgr_open) error: failed to import cht-bsw-rt5672 use case configuration -22 alsaucm: error failed to open sound card cht-bsw-rt5672: Invalid argument Replacing it with "RegexMatch" fixes this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/cht-bsw-rt5672/HiFi.conf | 4 ++-- ucm2/chtnau8824/HiFi.conf | 2 +- ucm2/chtrt5645/HiFi.conf | 6 +++--- ucm2/sof-soundwire/sof-soundwire.conf | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/ucm2/cht-bsw-rt5672/HiFi.conf b/ucm2/cht-bsw-rt5672/HiFi.conf index 394e45d..dc8ddd5 100644 --- a/ucm2/cht-bsw-rt5672/HiFi.conf +++ b/ucm2/cht-bsw-rt5672/HiFi.conf @@ -9,7 +9,7 @@ Define.HeadsetMic "yes" If.cfg-dmic1 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(SoMeThInK1)" # fixme! } @@ -21,7 +21,7 @@ If.cfg-dmic1 { If.cfg-dmic2 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(LENOVO.*ThinkPad8)" } diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf index 6b7624d..3ccd6b1 100644 --- a/ucm2/chtnau8824/HiFi.conf +++ b/ucm2/chtnau8824/HiFi.conf @@ -2,7 +2,7 @@ Define.Speaker "Speaker" If.cfg-mspk { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(cube-i1_TF|PIPO-W2S)" } diff --git a/ucm2/chtrt5645/HiFi.conf b/ucm2/chtrt5645/HiFi.conf index fb8636d..5d0e3bf 100644 --- a/ucm2/chtrt5645/HiFi.conf +++ b/ucm2/chtrt5645/HiFi.conf @@ -4,7 +4,7 @@ Define.DigitalMic "" If.cfg-dmic1 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(ASUSTeK.*T100HAN|ASUSTeK.*T101HA)" } @@ -16,7 +16,7 @@ If.cfg-dmic1 { If.cfg-dmic2 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(LENOVO.*LenovoMIIX320|MEDION.*Wingman)" } @@ -28,7 +28,7 @@ If.cfg-dmic2 { If.cfg-mspk { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(gpd-win-pocket-rt5645|TECLAST-X80Pro)" } diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index fe43d5f..2ba44be 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -38,7 +38,7 @@ DefineRegex { If.hs_init { Condition { - Type Regex + Type RegexMatch Regex "(rt5682|rt700|rt711)" String "${var:HeadsetCodec1}" } @@ -47,7 +47,7 @@ If.hs_init { If.mic_init { Condition { - Type Regex + Type RegexMatch Regex "(rt715)" String "${var:MicCodec1}" } -- 2.26.2

2 5

[PATCH] ucm: Allow empty strings in "${var:...}" substitutions
by Hans de Goede 28 Jun '20

28 Jun '20

Recent ucm-conf changes introduce checks like this one in various places: If.mspk { Condition { Type String Empty "${var:MonoSpeaker}" } True ... False ... } The 'Empty "${var:MonoSpeaker}"' part can only every succeed if we do: Define.MonoSpeaker "" But so far that would result in an error like this one: ALSA lib ucm_subs.c:367:(uc_mgr_get_substituted_value) variable '${var:MonoSpeaker}' is not defined in this context! ALSA lib main.c:983:(snd_use_case_mgr_open) error: failed to import cht-bsw-rt5672 use case configuration -22 alsaucm: error failed to open sound card cht-bsw-rt5672: Invalid argument This commit fixes this by allowing empty values for "${var:...}" substitutions. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Note besides the mentioned error, this also fixes similar errors I have been seeing on every board since alsa-ucm-conf commit d001c8de287f ("ucm.conf: add support for the kernel module name tree") --- src/ucm/ucm_subs.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/src/ucm/ucm_subs.c b/src/ucm/ucm_subs.c index 293426f2..a154aa51 100644 --- a/src/ucm/ucm_subs.c +++ b/src/ucm/ucm_subs.c @@ -262,9 +262,10 @@ static char *rval_var(snd_use_case_mgr_t *uc_mgr, const char *id) goto __rval; \ } -#define MATCH_VARIABLE2(name, id, fcn) \ +#define MATCH_VARIABLE2(name, id, fcn, empty_ok) \ if (strncmp((name), (id), sizeof(id) - 1) == 0) { \ idsize = sizeof(id) - 1; \ + allow_empty = (empty_ok); \ fcn2 = (fcn); \ goto __match2; \ } @@ -314,11 +315,11 @@ __std: MATCH_VARIABLE(value, "${CardName}", rval_card_name, false); MATCH_VARIABLE(value, "${CardLongName}", rval_card_longname, false); MATCH_VARIABLE(value, "${CardComponents}", rval_card_components, true); - MATCH_VARIABLE2(value, "${env:", rval_env); - MATCH_VARIABLE2(value, "${sys:", rval_sysfs); - MATCH_VARIABLE2(value, "${var:", rval_var); - MATCH_VARIABLE2(value, "${CardNumberByName:", rval_card_number_by_name); - MATCH_VARIABLE2(value, "${CardIdByName:", rval_card_id_by_name); + MATCH_VARIABLE2(value, "${env:", rval_env, false); + MATCH_VARIABLE2(value, "${sys:", rval_sysfs, false); + MATCH_VARIABLE2(value, "${var:", rval_var, true); + MATCH_VARIABLE2(value, "${CardNumberByName:", rval_card_number_by_name, false); + MATCH_VARIABLE2(value, "${CardIdByName:", rval_card_id_by_name, false); __merr: err = -EINVAL; tmp = strchr(value, '}'); -- 2.26.2

3 2

[PATCH 1/6] ASoC: Intel: cht_bsw_rt5672: Change bus format to I2S 2 channel
by Hans de Goede 28 Jun '20

28 Jun '20

The default mode for SSP configuration is TDM 4 slot and so far we were using this for the bus format on cht-bsw-rt56732 boards. One board, the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The second piggy-backed, output-only codec is inside the keyboard-dock (which has extra speakers). Unlike the main rt5672 codec, we cannot configure this codec, it is hard coded to use 2 channel 24 bit I2S. Using 4 channel TDM leads to the dock speakers codec (which listens in on the data send from the SSP to the rt5672 codec) emiting horribly distorted sound. Since we only support 2 channels anyways, there is no need for TDM on any cht-bsw-rt5672 designs. So we can simply use I2S 2ch everywhere. This commit fixes the Lenovo Miix 2 10 dock speakers issue by changing the bus format set in cht_codec_fixup() to I2S 2 channel. This change has been tested on the following devices with a rt5672 codec: Lenovo Miix 2 10 Lenovo Thinkpad 8 Lenovo Thinkpad 10 (gen 1) Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 7a43c70a1378..22e432768edb 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,21 +253,20 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot + * Default mode for SSP configuration is TDM 4 slot. One board/design, + * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The + * second piggy-backed, output-only codec is inside the keyboard-dock + * (which has extra speakers). Unlike the main rt5672 codec, we cannot + * configure this codec, it is hard coded to use 2 channel 24 bit I2S. + * Since we only support 2 channels anyways, there is no need for TDM + * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. */ - ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_IB_NF | + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) { - dev_err(rtd->dev, "can't set format to TDM %d\n", ret); - return ret; - } - - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; } -- 2.26.2

1 2

problem with Jabra EVOLVE 20 headset, and usb quirk update
by Hin-Tak Leung 27 Jun '20

27 Jun '20

Hi, This is a headset i.e. with both a ear set, and a microphone - the playback seems fine, but recording only accept one setting - and also I was having problem with zoom using it ("cannot get freq" see the tail end of this e-mail). switching via the gnome setting -> sound (from built-in hda) seems sluggish, and occasionally the volume is frozen to mute and unresponsive to unmute. $ arecord -f S16_LE -D front:CARD=J20,DEV=0 -d 10 wonder.wav Recording WAVE 'wonder.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Warning: rate is not accurate (requested = 8000Hz, got = 16000Hz) please, try the plug plugin and, also zoom seems to have problem getting at the microphone. I read that it is something to do with rate querying? Attached is "lsusb -vvvv -d 0b0e:0301 > usb-audio-lsusb-vvvv-0b0e:0301". I have tried just letting udev doing its thing, or preloading the module with with modprobe snd_usb_audio quirk_alias=0b0e0301:0b0e0349 or modprobe snd_usb_audio quirk_alias=0b0e0301:0b0e030b as below. But the "cannot get freq" message stays. This is based on similar vendor ids - sound/usb/format.c: if (chip->usb_id == USB_ID(0x0b0e, 0x030b)) { sound/usb/quirks.c: chip->usb_id == USB_ID(0x0b0e, 0x0349) || The device does not seems to be under quirks.c:snd_usb_ctl_msg_quirk() ( I don't really get delays, I think), format.c:formatparse_audio_format_rates_v1() seems to be somewhat related, but I don't have issue with playback, just capture. Should I insert this vid/pid into bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) as suggested elsewhere? I am just uncomfortable doing so as there isn't a quirk for this vendor there, yet. Also, related question: the format rate quirk routine and usb_ctl_msg_quirk routine does not seem to be affected by the alias_quirk list, as far as I understand and read the code? Here is the dmesg, under plain udev autoprobe, and the two preloads. I hope this is enough info. Please feel free to ask for more. (I also maintain https://github.com/HinTak/sound-usb-dkms/ as part of my previous job, but this device/usage is slightly complicated as I need to sign modified modules due to UEFI ; so I'd rather try something that has a chance of working, then just blindly try any modification) [25475.773095] usbcore: registered new interface driver snd-usb-audio [25482.559878] usb 2-2: new full-speed USB device number 6 using xhci_hcd [25482.764666] usb 2-2: New USB device found, idVendor=0b0e, idProduct=0301, bcdDevice= 3.00 [25482.764669] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [25482.764671] usb 2-2: Product: Jabra EVOLVE 20 [25482.764672] usb 2-2: Manufacturer: GN Netcom A/S [25482.764674] usb 2-2: SerialNumber: 0001B7518B0E07 [25482.770770] usb 2-2: device (0b0e:0301): applying quirk alias 0b0e:0349 [25482.794845] usb 2-2: 1:1: cannot get freq at ep 0x83 [25482.808838] usb 2-2: 2:1: cannot get freq at ep 0x4 [25483.133618] input: GN Netcom A/S Jabra EVOLVE 20 as /devices/pci0000:00/0000:00:10.0/usb2/2-2/2-2:1.3/0003:0B0E:0301.0004/input/input16 [25483.187155] jabra 0003:0B0E:0301.0004: input,hiddev96,hidraw0: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20] on usb-0000:00:10.0-2/input3 [25483.444718] usb 2-2: 2:1: cannot get freq at ep 0x4 [25865.406518] usb 2-2: USB disconnect, device number 6 [25883.242957] usbcore: deregistering interface driver snd-usb-audio [25895.379533] usb 2-2: new full-speed USB device number 7 using xhci_hcd [25895.584862] usb 2-2: New USB device found, idVendor=0b0e, idProduct=0301, bcdDevice= 3.00 [25895.584868] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [25895.584871] usb 2-2: Product: Jabra EVOLVE 20 [25895.584874] usb 2-2: Manufacturer: GN Netcom A/S [25895.584876] usb 2-2: SerialNumber: 0001B7518B0E07 [25895.640239] input: GN Netcom A/S Jabra EVOLVE 20 as /devices/pci0000:00/0000:00:10.0/usb2/2-2/2-2:1.3/0003:0B0E:0301.0005/input/input17 [25895.692123] jabra 0003:0B0E:0301.0005: input,hiddev96,hidraw0: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20] on usb-0000:00:10.0-2/input3 [25895.875421] usb 2-2: 1:1: cannot get freq at ep 0x83 [25895.885726] usb 2-2: 2:1: cannot get freq at ep 0x4 [25896.060320] usbcore: registered new interface driver snd-usb-audio [25896.149752] usb 2-2: 2:1: cannot get freq at ep 0x4 [25899.451644] usb 2-2: USB disconnect, device number 7 [25919.315839] usbcore: registered new interface driver snd-usb-audio [25924.595119] usb 2-2: new full-speed USB device number 8 using xhci_hcd [25924.797720] usb 2-2: New USB device found, idVendor=0b0e, idProduct=0301, bcdDevice= 3.00 [25924.797722] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [25924.797724] usb 2-2: Product: Jabra EVOLVE 20 [25924.797725] usb 2-2: Manufacturer: GN Netcom A/S [25924.797726] usb 2-2: SerialNumber: 0001B7518B0E07 [25924.802782] usb 2-2: device (0b0e:0301): applying quirk alias 0b0e:030b [25924.822715] usb 2-2: 1:1: cannot get freq at ep 0x83 [25924.832718] usb 2-2: 2:1: cannot get freq at ep 0x4 [25925.052655] input: GN Netcom A/S Jabra EVOLVE 20 as /devices/pci0000:00/0000:00:10.0/usb2/2-2/2-2:1.3/0003:0B0E:0301.0006/input/input18 [25925.104757] jabra 0003:0B0E:0301.0006: input,hiddev96,hidraw0: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20] on usb-0000:00:10.0-2/input3 [25925.338094] usb 2-2: 2:1: cannot get freq at ep 0x4 [26253.731814] usb 2-2: USB disconnect, device number 8

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[PATCH 16/17] ASoC: Intel: sdw_max98373: add card_late_probe support
by Pierre-Louis Bossart 26 Jun '20

26 Jun '20

From: randerwang <rander.wang(a)linux.intel.com> Disable Left and Right Spk pin after boot so that sof can get suspended. Please check patch: ASoC: Intel: Boards: tgl_max98373: add dai_trigger function Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: randerwang <rander.wang(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 19 ++++++++++++++++++- sound/soc/intel/boards/sof_sdw_common.h | 6 ++++++ sound/soc/intel/boards/sof_sdw_max98373.c | 12 ++++++++++++ 3 files changed, 36 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 45be9ec6d4ef..be8eccb50450 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -237,6 +237,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .direction = {true, true}, .dai_name = "max98373-aif1", .init = sof_sdw_mx8373_init, + .codec_card_late_probe = sof_sdw_mx8373_late_probe, }, { .id = 0x5682, @@ -927,13 +928,29 @@ static int sof_card_dai_links_create(struct device *dev, return 0; } +static int sof_sdw_card_late_probe(struct snd_soc_card *card) +{ + int i, ret; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (!codec_info_list[i].late_probe) + continue; + + ret = codec_info_list[i].codec_card_late_probe(card); + if (ret < 0) + return ret; + } + + return sof_sdw_hdmi_card_late_probe(card); +} + /* SoC card */ static const char sdw_card_long_name[] = "Intel Soundwire SOF"; static struct snd_soc_card card_sof_sdw = { .name = "soundwire", .owner = THIS_MODULE, - .late_probe = sof_sdw_hdmi_card_late_probe, + .late_probe = sof_sdw_card_late_probe, .codec_conf = codec_conf, .num_configs = ARRAY_SIZE(codec_conf), }; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 3f820cf99a89..426017626b16 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -11,6 +11,7 @@ #include <linux/bits.h> #include <linux/types.h> +#include <sound/soc.h> #define MAX_NO_PROPS 2 #define MAX_HDMI_NUM 4 @@ -61,6 +62,9 @@ struct sof_sdw_codec_info { struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); + + bool late_probe; + int (*codec_card_late_probe)(struct snd_soc_card *card); }; struct mc_private { @@ -114,6 +118,8 @@ int sof_sdw_mx8373_init(const struct snd_soc_acpi_link_adr *link, struct sof_sdw_codec_info *info, bool playback); +int sof_sdw_mx8373_late_probe(struct snd_soc_card *card); + /* RT5682 support */ int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c index a38ddc099a95..6437872a9b3d 100644 --- a/sound/soc/intel/boards/sof_sdw_max98373.c +++ b/sound/soc/intel/boards/sof_sdw_max98373.c @@ -68,7 +68,19 @@ int sof_sdw_mx8373_init(const struct snd_soc_acpi_link_adr *link, if (info->amp_num == 2) dai_links->init = spk_init; + info->late_probe = true; + dai_links->ops = &max_98373_sdw_ops; return 0; } + +int sof_sdw_mx8373_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dapm_context *dapm = &card->dapm; + + /* Disable Left and Right Spk pin after boot */ + snd_soc_dapm_disable_pin(dapm, "Left Spk"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + return snd_soc_dapm_sync(dapm); +} -- 2.20.1

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Fwd: [PATCH] snd/soc: correct trivial kernel-doc inconsistency
by Colton Lewis 26 Jun '20

26 Jun '20

Can some please accept or comment on this patch? ---------- Forwarded Message ---------- Subject: [PATCH] snd/soc: correct trivial kernel-doc inconsistency Date: Saturday, June 13, 2020, 9:29:08 PM CDT From: Colton Lewis <colton.w.lewis(a)protonmail.com> To: lgirdwood(a)gmail.com CC: trivial(a)kernel.org, Colton Lewis <colton.w.lewis(a)protonmail.com> Silence documentation build warning by correcting kernel-doc comment for snd_soc_runtime_action. ./sound/soc/soc-pcm.c:220: warning: Function parameter or member 'action' not described in 'snd_soc_runtime_action' Signed-off-by: Colton Lewis <colton.w.lewis(a)protonmail.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c114b4542ce..9ab376844ac1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -208,6 +208,7 @@ static inline void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm) * PCM runtime components * @rtd: ASoC PCM runtime that is activated * @stream: Direction of the PCM stream + * @action: Activate stream if 1. Deactivate if -1. * * Increments/Decrements the active count for all the DAIs and components * attached to a PCM runtime. -- 2.26.2 -----------------------------------------

3 4

Cannot use HyperX Cloud Flight S Microphone
by GitHub issues - opened 25 Jun '20

25 Jun '20

alsa-project/alsa-lib issue #65 was opened from DirtyCajunRice: I have a New USB headset that was added/fixed in kernel patch 5.7rc3 (https://lkml.org/lkml/2020/4/4/197). It now loads all the output options but no input options. Is there something i can do to force it? also have this if it helps: http://alsa-project.org/db/?f=7cf962fe3df280a9a81f5cd1e82633d8cba7b02e Issue URL : https://github.com/alsa-project/alsa-lib/issues/65 Repository URL: https://github.com/alsa-project/alsa-lib

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