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

[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

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[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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[PATCH 00/10] Audio graph card updates and usage with Tegra210 audio
by Sameer Pujar 03 Aug '20

03 Aug '20

This series proposes following enhancements to audio-graph card driver. * Support multiple instances of a component. * Support Codec port with multiple endpoints. * Support open platforms with empty Codec endpoint. * Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Add new compatible to support DPCM based DAI chaining. This pushes DT support for Tegra210 based platforms which uses audio-graph card and above enhancements. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Sameer Pujar (10): ASoC: soc-core: Fix component name_prefix parsing ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Support Codec with multiple endpoints ASoC: audio-graph: Support for empty end point ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: soc-pcm: Get all BEs along DAPM path ASoC: dt-bindings: audio-graph-card: Support for component chaining ASoC: audio-graph: Add support for component chaining arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../devicetree/bindings/sound/audio-graph-card.txt | 1 + .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 120 +++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 196 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 110 ++++++++++++ include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 85 ++++++--- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- 8 files changed, 495 insertions(+), 24 deletions(-) create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi -- 2.7.4

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[PATCH] ASoC: wm8962: Do not access WM8962_GPIO_BASE
by Fabio Estevam 03 Aug '20

03 Aug '20

According to the WM8962 datasheet, there is no register at address 0x200. WM8962_GPIO_BASE is just a base address for the GPIO registers and not a real register, so remove it from wm8962_readable_register(). Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip its access. This fixes the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Hi, There is still one more soc_component_read_no_lock error left on register 48. I can get rid of it with the below change: --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -151,6 +151,7 @@ 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 */ @@ -841,7 +842,6 @@ 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: I haven't submitted it yet because I don't know if this is the correct approach. Please advise. Thanks sound/soc/codecs/wm8962.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index df8cdc71357d..8159a3866cde 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -956,7 +956,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_EQ39: case WM8962_EQ40: case WM8962_EQ41: - case WM8962_GPIO_BASE: case WM8962_GPIO_2: case WM8962_GPIO_3: case WM8962_GPIO_5: @@ -3437,7 +3436,13 @@ static int wm8962_probe(struct snd_soc_component *component) /* Save boards having to disable DMIC when not in use */ dmicclk = false; dmicdat = false; - for (i = 0; i < WM8962_MAX_GPIO; i++) { + for (i = 1; i < WM8962_MAX_GPIO; i++) { + /* + * Register 515 (WM8962_GPIO_BASE + 3) does not exist, + * so skip its access + */ + if (i == 3) + continue; switch (snd_soc_component_read(component, WM8962_GPIO_BASE + i) & WM8962_GP2_FN_MASK) { case WM8962_GPIO_FN_DMICCLK: -- 2.17.1

4 9

[PATCH v3 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 03 Aug '20

03 Aug '20

Note: - The machine driver patch depends on LPASS patch series so it is not ready to be merged now. ASoC: qcom: Add support for SC7180 lpass variant https://patchwork.kernel.org/cover/11678133/ - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Thanks for the review! Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 113 ++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 244 ++++++++++++++++++ 4 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.163.g6104cc2f0b6-goog

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[PATCH v2] 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 v2 - replace is_stream_in_use with streams - add "out" error handler in hw_params() sound/soc/fsl/fsl-asoc-card.c | 155 ++++++++++++++++------------------ 1 file changed, 71 insertions(+), 84 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4848ba61d083..b6a2a527dc04 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 out; } if (cpu_priv->slot_width) { @@ -182,6 +180,69 @@ 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 out; + } + } + + /* 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 out; + } + + 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 out; + } + } + + return 0; + +out: + 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 +252,7 @@ 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 +316,6 @@ 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 +604,6 @@ 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 +620,22 @@ 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 +645,6 @@ 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

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[PATCH] ALSA: docs: fix typo
by Julia Lawall 01 Aug '20

01 Aug '20

GFP_KRENEL -> GFP_KERNEL Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index fa49688..aa9d5ab 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3579,7 +3579,7 @@ dependent on the bus. For normal devices, pass the device pointer ``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type. You can pass NULL to the device pointer in that case, which is the -default mode implying to allocate with ``GFP_KRENEL`` flag. +default mode implying to allocate with ``GFP_KERNEL`` flag. If you need a different GFP flag, you can pass it by encoding the flag into the device pointer via a special macro :c:func:`snd_dma_continuous_data()`.

2 1

[PATCH] ALSA: doc: use correct config variable name
by Julia Lawall 01 Aug '20

01 Aug '20

CONFIG_PCM_XRUN_DEBUG should be CONFIG_SND_PCM_XRUN_DEBUG Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- Documentation/sound/designs/procfile.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/designs/procfile.rst b/Documentation/sound/designs/procfile.rst index 29a4668..e9f7e0c 100644 --- a/Documentation/sound/designs/procfile.rst +++ b/Documentation/sound/designs/procfile.rst @@ -91,7 +91,7 @@ PCM Proc Files ``card*/pcm*/xrun_debug`` This file appears when ``CONFIG_SND_DEBUG=y`` and - ``CONFIG_PCM_XRUN_DEBUG=y``. + ``CONFIG_SND_PCM_XRUN_DEBUG=y``. This shows the status of xrun (= buffer overrun/xrun) and invalid PCM position debug/check of ALSA PCM middle layer. It takes an integer value, can be changed by writing to this

2 1

[PATCH v2 1/2] dt-bindings: sound: add DT bindings for Microchip S/PDIF TX Controller
by Codrin Ciubotariu 31 Jul '20

31 Jul '20

This patch adds DT bindings for the new Microchip S/PDIF TX Controller embedded inside sama7g5 SoCs. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v2: - replaced https with http - reworked example, included bindings; .../bindings/sound/mchp,spdiftx.yaml | 80 +++++++++++++++++++ 1 file changed, 80 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml new file mode 100644 index 000000000000..2b7c27391f0d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller Device Tree Bindings + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a + serial port compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + + dma-names: + const: tx + + pinctrl-0: true + + pinctrl-names: + const: default + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x4000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; -- 2.25.1

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[PATCH 0/3] ASoC: core: Two step component registration
by Cezary Rojewski 31 Jul '20

31 Jul '20

Provide a mechanism for true two-step component registration. This mimics device registration flow where initialization is the first step while addition goes as second in line. Drivers may choose to modify component's fields before registering component to ASoC subsystem via snd_soc_add_component. Patchset achieves status quo - behavior of snd_soc_register_component remains unchanged. Cezary Rojewski (3): ASoC: core: Relocate and expose snd_soc_component_initialize ASoC: core: Simplify snd_soc_component_initialize declaration ASoC: core: Two step component registration include/sound/soc-component.h | 3 -- include/sound/soc.h | 11 +++--- sound/soc/soc-component.c | 16 --------- sound/soc/soc-core.c | 52 +++++++++++++++++---------- sound/soc/soc-generic-dmaengine-pcm.c | 14 +++++--- sound/soc/stm/stm32_adfsdm.c | 9 +++-- 6 files changed, 55 insertions(+), 50 deletions(-) -- 2.17.1

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