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[PATCH] ALSA: hda/hdmi: Add pins with jack detection support
by Kai-Heng Feng 04 Aug '20

04 Aug '20

HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 Those pins were filtered out by commit 116dcde63806 ("ALSA: HDA: Remove unconnected PCM devices for Intel HDMI"). However, jacks that support detection won't have the issues the commit addresses. So still add the pin if it supports jack detection. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cd46247988e4..db3a5148bd40 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1701,7 +1701,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if ((get_defcfg_connect(config) == AC_JACK_PORT_NONE) && + !(caps & AC_PINCAP_PRES_DETECT)) return 0; /* -- 2.17.1

2 5

[PATCH 0/3] ASoC: codec: tlv3204: Codec workaround
by Michael Sit Wei Hong 04 Aug '20

04 Aug '20

This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing v1: initial ASoC: codec: tlv3204: Codec workaround series Michael Sit Wei Hong (3): ASoC: codec: tlv3204: Enable 24 bit audio support ASoC: codec: tlv3204: Increased maximum supported channels ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset sound/soc/codecs/tlv320aic32x4.c | 60 +++++++++++++++++++++++--------- 1 file changed, 44 insertions(+), 16 deletions(-) -- 2.17.1

5 10

[PATCH] ASoC: fsl_sai: Clean code for synchronize mode
by Shengjiu Wang 04 Aug '20

04 Aug '20

TX synchronous with RX: The RMR is no need to be changed when Tx is enabled, the other configuration in hw_params() is enough for clock generation. The TCSR.TE is no need to enabled when only RX is enabled. RX synchronous with TX: The TMR is no need to be changed when Rx is enabled, the other configuration in hw_params() is enough for clock generation. The RCSR.RE is no need to enabled when only TX is enabled. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdff739924e2..a210c9836a9a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -482,8 +482,6 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_TCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); - regmap_write(sai->regmap, FSL_SAI_TMR, - ~0UL - ((1 << channels) - 1)); } else if (!sai->synchronous[RX] && sai->synchronous[TX] && tx) { regmap_update_bits(sai->regmap, FSL_SAI_RCR4(ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, @@ -491,8 +489,6 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_RCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); - regmap_write(sai->regmap, FSL_SAI_RMR, - ~0UL - ((1 << channels) - 1)); } } @@ -553,11 +549,18 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs), - FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs), + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + /* Enable opposite direction when necessarily */ + if (!sai->synchronous[TX] && sai->synchronous[RX] && !tx) { + regmap_update_bits(sai->regmap, FSL_SAI_xCSR((!tx), ofs), + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + } else if (!sai->synchronous[RX] && sai->synchronous[TX] && tx) { + regmap_update_bits(sai->regmap, FSL_SAI_xCSR((!tx), ofs), + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + } + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); break; -- 2.27.0

3 13

[PATCH] snd: usb: add startech usb audio dock name
by Curtis Malainey 04 Aug '20

04 Aug '20

The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name and shows up currently as "USB Sound Device" in ALSA. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/usb/quirks-table.h | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9092cc0aa8072..fef06836115c1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3252,11 +3252,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ +{ + USB_DEVICE(0x0d8c, 0x0102), + QUIRK_RENAME_DEVICE(NULL, "ICUSBAUDIO7D") +}, { - /* - * The original product_name is "USB Sound Device", however this name - * is also used by the CM106 based cards, so make it unique. - */ USB_DEVICE(0x0d8c, 0x0103), QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII") }, -- 2.28.0.163.g6104cc2f0b6-goog

2 1

[PATCH] Revert "ALSA: hda: call runtime_allow() for all hda controllers"
by Hui Wang 04 Aug '20

04 Aug '20

This reverts commit 9a6418487b56 ("ALSA: hda: call runtime_allow() for all hda controllers"). The reverted patch already introduced some regressions on some machines: - on gemini-lake machines, the error of "azx_get_response timeout" happens in the hda driver. - on the machines with alc662 codec, the audio jack detection doesn't work anymore. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e699873c8293..e34a4d5d047c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2352,7 +2352,6 @@ 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

3 7

[PATCH 0/5] ASoC: tegra: Fix compile warning with CONFIG_PM=n
by Takashi Iwai 03 Aug '20

03 Aug '20

Hi, this is a trivial patch set to add the missing __maybe_unused for covering the compile warnings with CONFIG_PM=n. Takashi === Takashi Iwai (5): ASoC: tegra: tegra186_dspk: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_admaif: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n sound/soc/tegra/tegra186_dspk.c | 4 ++-- sound/soc/tegra/tegra210_admaif.c | 4 ++-- sound/soc/tegra/tegra210_ahub.c | 4 ++-- sound/soc/tegra/tegra210_dmic.c | 4 ++-- sound/soc/tegra/tegra210_i2s.c | 4 ++-- 5 files changed, 10 insertions(+), 10 deletions(-) -- 2.16.4

3 7

[PATCH v3] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
by Shengjiu Wang 03 Aug '20

03 Aug '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> --- changes in v3 - replace "out" with "fail" changes in v2 - replace is_stream_in_use with streams - add "out" error handler in hw_params() sound/soc/fsl/fsl-asoc-card.c | 154 ++++++++++++++++------------------ 1 file changed, 70 insertions(+), 84 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index ee80d02b56c6..52a9fc73ceb8 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 + * @streams: Mask of current active streams * @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; + u8 streams; 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->streams |= BIT(substream->stream); - /* - * 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 */ @@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); - return ret; + goto fail; } if (cpu_priv->slot_width) { @@ -182,6 +180,68 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); + goto fail; + } + } + + /* 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); + goto fail; + } + + 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); + goto fail; + } + } + + return 0; + +fail: + priv->streams &= ~BIT(substream->stream); + return ret; +} + +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); + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->streams &= ~BIT(substream->stream); + + if (!priv->streams && 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; } } @@ -191,6 +251,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, 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 +315,6 @@ 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) { @@ -608,7 +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; @@ -625,26 +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; @@ -654,7 +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 2

[PATCH] ASoC: wm8962: Do not remove ADDITIONAL_CONTROL_4 from readable register list
by Fabio Estevam 03 Aug '20

03 Aug '20

Removing ADDITIONAL_CONTROL_4 from the list of readable registers cause audio distortion. This change was sent as a comment below the --- line when submitting commit 658bb297e393 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE"), so it was not supposed to get merged. Keep WM8962_ADDITIONAL_CONTROL_4 inside wm8962_readable_register() to fix the regression. Fixes: 658bb297e393 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE") Reported-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 317916cb4e27..0623a2251084 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -151,7 +151,6 @@ static const struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ - { 48, 0x0000 }, /* R48 - Additional control(4) */ { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ @@ -842,6 +841,7 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_SPKOUTL_VOLUME: case WM8962_SPKOUTR_VOLUME: case WM8962_THERMAL_SHUTDOWN_STATUS: + case WM8962_ADDITIONAL_CONTROL_4: case WM8962_CLASS_D_CONTROL_1: case WM8962_CLASS_D_CONTROL_2: case WM8962_CLOCKING_4: -- 2.17.1

2 2

[PATCH] ASoC: amd: renoir: restore two more registers during resume
by Hui Wang 03 Aug '20

03 Aug '20

Recently we found an issue about the suspend and resume. If dmic is recording the sound, and we run suspend and resume, after the resume, the dmic can't work well anymore. we need to close the app and reopen the app, then the dmic could record the sound again. For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2 -r 48000 test.wav", then suspend and resume, after the system resume back, we speak to the dmic. then stop the arecord, use aplay to play the test.wav, we could hear the sound recorded after resume is weird, it is not what we speak to the dmic. I found two registers are set in the dai_hw_params(), if the two registers are set in the resume() too, this issue could be fixed. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 623dfd3ea705..8acb0315a169 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -474,6 +474,11 @@ static int acp_pdm_resume(struct device *dev) rtd = runtime->private_data; period_bytes = frames_to_bytes(runtime, runtime->period_size); buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + if (runtime->channels == TWO_CH) { + rn_writel(0x0 , rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); + rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + + ACP_WOV_PDM_DECIMATION_FACTOR); + } config_acp_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); init_pdm_ring_buffer(MEM_WINDOW_START, buffer_len, period_bytes, adata->acp_base); -- 2.17.1

3 3

[PATCH v2] ASoC: amd: renoir: restore two more registers during resume
by Hui Wang 03 Aug '20

03 Aug '20

Recently we found an issue about the suspend and resume. If dmic is recording the sound, and we run suspend and resume, after the resume, the dmic can't work well anymore. we need to close the app and reopen the app, then the dmic could record the sound again. For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2 -r 48000 test.wav", then suspend and resume, after the system resume back, we speak to the dmic. then stop the arecord, use aplay to play the test.wav, we could hear the sound recorded after resume is weird, it is not what we speak to the dmic. I found two registers are set in the dai_hw_params(), if the two registers are set during the resume, this issue could be fixed. Move the code of the dai_hw_params() into the pdm_dai_trigger(), then these two registers will be set during resume since pdm_dai_trigger() will be called during resume. And delete the empty function dai_hw_params(). Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 29 +++++++++------------------- 1 file changed, 9 insertions(+), 20 deletions(-) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 623dfd3ea705..7b14d9a81b97 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -314,40 +314,30 @@ static int acp_pdm_dma_close(struct snd_soc_component *component, return 0; } -static int acp_pdm_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) { struct pdm_stream_instance *rtd; + int ret; + bool pdm_status; unsigned int ch_mask; rtd = substream->runtime->private_data; - switch (params_channels(params)) { + ret = 0; + switch (substream->runtime->channels) { case TWO_CH: ch_mask = 0x00; break; default: return -EINVAL; } - rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); - rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + - ACP_WOV_PDM_DECIMATION_FACTOR); - return 0; -} - -static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct pdm_stream_instance *rtd; - int ret; - bool pdm_status; - - rtd = substream->runtime->private_data; - ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); + rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + + ACP_WOV_PDM_DECIMATION_FACTOR); rtd->bytescount = acp_pdm_get_byte_count(rtd, substream->stream); pdm_status = check_pdm_dma_status(rtd->acp_base); @@ -369,7 +359,6 @@ static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, } static struct snd_soc_dai_ops acp_pdm_dai_ops = { - .hw_params = acp_pdm_dai_hw_params, .trigger = acp_pdm_dai_trigger, }; -- 2.17.1

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