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[PATCH 1/2] ASoC: tlv320adcx140: Fix GPO register start address
by Dan Murphy 31 Jul '20

31 Jul '20

The header was updated to align with the data sheet to start the GPO_CFG at GPO_CFG0. The code was not updated to the change and therefore the GPO_CFG0 register was not written to. Fixes: 6617cff6a05e ("ASoC: tlv320adcx140: Add GPO configuration and drive output config") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 7bb9e9aa0c0a..fc7616dcf9b5 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -790,7 +790,7 @@ static int adcx140_configure_gpo(struct adcx140_priv *adcx140) gpo_output_val = gpo_outputs[0] << ADCX140_GPO_SHIFT | gpo_outputs[1]; - ret = regmap_write(adcx140->regmap, ADCX140_GPO_CFG1 + i, + ret = regmap_write(adcx140->regmap, ADCX140_GPO_CFG0 + i, gpo_output_val); if (ret) return ret; -- 2.28.0

2 2

[PATCH v2 1/5] ASoC: amd: Renaming snd-soc-card structure and fields
by Ravulapati Vishnu vardhan rao 31 Jul '20

31 Jul '20

As in future our machine driver supports multiple codecs So changing naming convention of snd_soc_card struct and its fields. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 6009e444b858..8b5af064864f 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -290,7 +290,7 @@ static const struct snd_kcontrol_new acp3x_dmic_mux_control = SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum, dmic_get, dmic_set); -static const struct snd_soc_dapm_widget acp3x_widgets[] = { +static const struct snd_soc_dapm_widget acp3x_5682_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -298,7 +298,7 @@ static const struct snd_soc_dapm_widget acp3x_widgets[] = { &acp3x_dmic_mux_control), }; -static const struct snd_soc_dapm_route acp3x_audio_route[] = { +static const struct snd_soc_dapm_route acp3x_5682_audio_route[] = { {"Headphone Jack", NULL, "HPOL"}, {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, @@ -307,23 +307,23 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Dmic Mux", "Rear Mic", "DMIC"}, }; -static const struct snd_kcontrol_new acp3x_mc_controls[] = { +static const struct snd_kcontrol_new acp3x_5682_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Spk"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; -static struct snd_soc_card acp3x_card = { +static struct snd_soc_card acp3x_5682 = { .name = "acp3xalc5682m98357", .owner = THIS_MODULE, .dai_link = acp3x_dai_5682_98357, .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), - .dapm_widgets = acp3x_widgets, - .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), - .dapm_routes = acp3x_audio_route, - .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), - .controls = acp3x_mc_controls, - .num_controls = ARRAY_SIZE(acp3x_mc_controls), + .dapm_widgets = acp3x_5682_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_5682_widgets), + .dapm_routes = acp3x_5682_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_5682_audio_route), + .controls = acp3x_5682_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; static int acp3x_probe(struct platform_device *pdev) @@ -336,8 +336,8 @@ static int acp3x_probe(struct platform_device *pdev) if (!machine) return -ENOMEM; - card = &acp3x_card; - acp3x_card.dev = &pdev->dev; + card = &acp3x_5682; + acp3x_5682.dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); @@ -348,11 +348,11 @@ static int acp3x_probe(struct platform_device *pdev) return PTR_ERR(dmic_sel); } - ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_5682); if (ret) { dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - acp3x_card.name, ret); + acp3x_5682.name, ret); return ret; } return 0; -- 2.17.1

2 1

[PATCH] ASoC: rk3399_gru_sound: Add DAPM pins, kcontrols for jack detection
by Alper Nebi Yasak 31 Jul '20

31 Jul '20

PulseAudio (and perhaps other userspace utilities) can not detect any jack for rk3399_gru_sound as the driver doesn't expose related Jack kcontrols. This patch adds two DAPM pins to the headset jack, where the snd_soc_card_jack_new() call automatically creates "Headphones Jack" and "Headset Mic Jack" kcontrols from them. With an appropriate ALSA UCM config specifying JackControl fields for the "Headphones" and "Headset" (mic) devices, PulseAudio can detect plug/unplug events for both of them after this patch. Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index f45e5aaa4b30..c17d146717a8 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -32,6 +32,19 @@ static unsigned int dmic_wakeup_delay; static struct snd_soc_jack rockchip_sound_jack; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin rockchip_sound_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + +}; + static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -176,7 +189,9 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - &rockchip_sound_jack, NULL, 0); + &rockchip_sound_jack, + rockchip_sound_jack_pins, + ARRAY_SIZE(rockchip_sound_jack_pins)); if (ret) { dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); -- 2.28.0.rc1

2 1

[PATCH] ASoC: da7219: Fix general protection fault in da7219_register_dai_clks
by Yong Zhi 30 Jul '20

30 Jul '20

clkdev_drop(cl) does not null the removed cl, if da7219_register_dai_clks() entered again after card removal, devm_clk_register() will return -EEXIST, the goto err to clkdev_drop() will trigger board reboot. Test commands: modprobe -r snd_sof_pci modprobe snd_sof_pci The oops looks like: da7219 i2c-DLGS7219:00: Using default DAI clk names: da7219-dai-wclk, da7219-dai-bclk da7219 i2c-DLGS7219:00: Failed to register da7219-dai-wclk: -17 general protection fault: 0000 [#1] PREEMPT SMP NOPTI RIP: 0010:clkdev_drop+0x20/0x52 Call Trace: da7219_probe+0x52e/0x6dc [snd_soc_da7219] soc_probe_component+0x206/0x3a1 snd_soc_bind_card+0x4ee/0x9a6 devm_snd_soc_register_card+0x48/0x7b audio_probe+0x1f0/0x221 [snd_soc_sof_da7219_max98373] platform_drv_probe+0x89/0xa2 really_probe+0x129/0x30d driver_probe_device+0x59/0xec ? driver_sysfs_remove+0x55/0x55 bus_for_each_drv+0xa1/0xdc __device_attach+0xc2/0x146 bus_probe_device+0x32/0x97 device_add+0x311/0x3b4 platform_device_add+0x184/0x1eb Fix by marking (nullifying) the da7219->dai_clks_lookup[i] after clkdev_drop(). Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/codecs/da7219.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 153ea30b5a8f..54da7cfbb5f4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2369,8 +2369,10 @@ static void da7219_remove(struct snd_soc_component *component) #ifdef CONFIG_COMMON_CLK for (i = DA7219_DAI_NUM_CLKS - 1; i >= 0; --i) { - if (da7219->dai_clks_lookup[i]) + if (da7219->dai_clks_lookup[i]) { clkdev_drop(da7219->dai_clks_lookup[i]); + da7219->dai_clks_lookup[i] = NULL; + } } #endif -- 2.7.4

5 7

[PATCH] ALSA: hda: fix NULL pointer dereference during suspend
by Ranjani Sridharan 30 Jul '20

30 Jul '20

When the ASoC card registration fails and the codec component driver never probes, the codec device is not initialized and therefore memory for codec->wcaps is not allocated. This results in a NULL pointer dereference when the codec driver suspend callback is invoked during system suspend. Fix this by returning without performing any actions during codec suspend/resume if the card was not registered successfully. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3576e2d8452f..9b1f387d18e5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2936,6 +2936,10 @@ static int hda_codec_runtime_suspend(struct device *dev) struct hda_codec *codec = dev_to_hda_codec(dev); unsigned int state; + /* Nothing to do if card registration fails and the component driver never probes */ + if (!codec->card) + return 0; + cancel_delayed_work_sync(&codec->jackpoll_work); state = hda_call_codec_suspend(codec); if (codec->link_down_at_suspend || @@ -2950,6 +2954,10 @@ static int hda_codec_runtime_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + /* Nothing to do if card registration fails and the component driver never probes */ + if (!codec->card) + return 0; + codec_display_power(codec, true); snd_hdac_codec_link_up(&codec->core); hda_call_codec_resume(codec); -- 2.25.1

3 6

[PATCH v2 0/2] Add period size constraint for Atom Chromebook
by Brent Lu 30 Jul '20

30 Jul '20

Two different constraints are implemented: one is in platform's CPU DAI to enforce period sizes which are already used in Android BSP. The other is in Atom Chromebook's machine driver to use 240 as period size. Changes since v1: -Add comma at the end of media_period_size array declaration. Brent Lu (1): ASoC: intel: atom: Add period size constraint Yu-Hsuan Hsu (1): ASoC: Intel: Add period size constraint on strago board sound/soc/intel/atom/sst-mfld-platform-pcm.c | 15 +++++++++++++++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 14 +++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++++++++++++- 3 files changed, 41 insertions(+), 2 deletions(-) -- 2.7.4

4 8

[PATCH][next] ALSA: vmaster: Use flex_array_size() helper in memcpy()
by Gustavo A. R. Silva 30 Jul '20

30 Jul '20

Make use of the flex_array_size() helper to calculate the size of a flexible array member within an enclosing structure. This helper offers defense-in-depth against potential integer overflows and makes it explicitly clear that we are dealing with a flexible array member. Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/core/vmaster.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index ab36f9898711..21ce4082cb5f 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -262,7 +262,8 @@ int _snd_ctl_add_follower(struct snd_kcontrol *master, return -ENOMEM; srec->kctl = follower; srec->follower = *follower; - memcpy(srec->follower.vd, follower->vd, follower->count * sizeof(*follower->vd)); + memcpy(srec->follower.vd, follower->vd, flex_array_size(srec, follower.vd, + srec->follower.count)); srec->master = master_link; srec->flags = flags; -- 2.27.0

3 3

HP NB right speaker no sound
by Kailang 30 Jul '20

30 Jul '20

Hi Takashi, HP NB platform right side speaker no sound. Attach patch was help EC to initial I2S Amp normally. Thanks. BR, Kailang

2 3

[PATCH] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
by Shengjiu Wang 30 Jul '20

30 Jul '20

With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 149 +++++++++++++++------------------- 1 file changed, 66 insertions(+), 83 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4848ba61d083..0517dbb3e908 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @is_stream_in_use: flags for release resource in hw_free * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + bool is_stream_in_use[2]; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->is_stream_in_use[tx] = true; - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -185,12 +183,72 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, return ret; } } + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->is_stream_in_use[tx] = false; + + if (!priv->is_stream_in_use[!tx] && codec_priv->pll_id && + codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + } return 0; } static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +312,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); - if (ret) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -611,7 +600,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -628,26 +616,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -657,7 +641,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; -- 2.27.0

3 4

Headset microphone not detected
by Ujjwal Sharma 30 Jul '20

30 Jul '20

Hi everyone! I am unable to get my headset microphone to work, and that's really puzzling to me since the headphone speakers show up as plugged and work perfectly (pulseaudio shows the headset microphone as unplugged). I reported this issue to the SOF issue tracker on GitHub and they pointed out that it's an ALSA issue. Here's the mentioned thread: https://github.com/thesofproject/sof/issues/3164. My alsa_info.sh: https://pastebin.com/raw/K99520LA Some snippets from there that might be interesting: [ 7612.883978] snd_hda_codec_realtek ehdaudio0D0: didn't find PCM for DAI Digital Codec DAI [ 7612.883986] snd_hda_codec_realtek ehdaudio0D0: ASoC: can't open DAI Digital Codec DAI: -22 [ 7612.883992] snd_hda_codec_realtek ehdaudio0D0: didn't find PCM for DAI Digital Codec DAI [ 7612.883998] Digital Playback and Capture: ASoC: BE open failed -22 [ 7612.884002] HDA Digital: ASoC: failed to start some BEs -22 state.sofhdadsp { .... name 'Headset Jack' value false ... name 'Mic Jack' value false I cannot possibly understand what's wrong. Is it a bug in the codecs? Thanks, Ujjwal

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