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[PATCH][next] ASoC: q6asm: use kcalloc() instead of kzalloc()
by Gustavo A. R. Silva 28 Jul '22

28 Jul '22

Use 2-factor multiplication argument form kcalloc() instead of kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 9251d8548965..195780f75d05 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -513,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, return 0; } - buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC); + buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC); if (!buf) { spin_unlock_irqrestore(&ac->lock, flags); return -ENOMEM; -- 2.27.0

3 2

[PATCH v2] ASoC: amd: acp: Initialize list to store acp_stream during pcm_open
by Venkata Prasad Potturu 28 Jul '22

28 Jul '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> We are currently allocating acp_stream during pcm_open and saving it in static array corresponds to array index calculated based on cpu dai->driver id. This approach will fail if we have single dai linked to multiple pcm device as we will have same dai->driver id or array index for multiple pcm open. Initialize new linked list stream_list to store opened pcm stream info dynamically. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-platform.c | 33 ++++++++++++-------------------- sound/soc/amd/acp/amd.h | 3 ++- 2 files changed, 14 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 10730d33c3b0..20c0f75f7c97 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -94,7 +94,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; u16 i2s_flag = 0; - u32 ext_intr_stat, ext_intr_stat1, i; + u32 ext_intr_stat, ext_intr_stat1; if (!adata) return IRQ_NONE; @@ -104,8 +104,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); - for (i = 0; i < ACP_MAX_STREAM; i++) { - stream = adata->stream[i]; + list_for_each_entry(stream, &adata->stream_list, list) { if (stream && (ext_intr_stat & stream->irq_bit)) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); @@ -146,9 +145,8 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL); } -static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) +static void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int size) { - struct acp_stream *stream = adata->stream[cpu_id]; struct snd_pcm_substream *substream = stream->substream; struct acp_resource *rsrc = adata->rsrc; dma_addr_t addr = substream->dma_buffer.addr; @@ -174,13 +172,10 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct snd_pcm_runtime *runtime = substream->runtime; struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; int ret; stream = kzalloc(sizeof(*stream), GFP_KERNEL); @@ -188,7 +183,8 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs return -ENOMEM; stream->substream = substream; - adata->stream[stream_id] = stream; + + list_add_tail(&stream->list, &adata->stream_list); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp_pcm_hardware_playback; @@ -212,16 +208,13 @@ static int acp_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); struct acp_dev_data *adata = snd_soc_component_get_drvdata(component); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct acp_stream *stream = substream->runtime->private_data; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; u64 size = params_buffer_bytes(params); /* Configure ACP DMA block with params */ config_pte_for_stream(adata, stream); - config_acp_dma(adata, stream_id, size); + config_acp_dma(adata, stream, size); return 0; } @@ -261,16 +254,11 @@ static int acp_dma_new(struct snd_soc_component *component, static int acp_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); - struct device *dev = component->dev; - struct acp_dev_data *adata = dev_get_drvdata(dev); - struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; + struct acp_stream *stream = substream->runtime->private_data; - stream = adata->stream[stream_id]; + /* Remove entry from list */ + list_del(&stream->list); kfree(stream); - adata->stream[stream_id] = NULL; return 0; } @@ -305,6 +293,9 @@ int acp_platform_register(struct device *dev) dev_err(dev, "Fail to register acp i2s component\n"); return status; } + + INIT_LIST_HEAD(&adata->stream_list); + return 0; } EXPORT_SYMBOL_NS_GPL(acp_platform_register, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..148a9ab6206d 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -91,6 +91,7 @@ struct acp_chip_info { }; struct acp_stream { + struct list_head list; struct snd_pcm_substream *substream; int irq_bit; int dai_id; @@ -123,7 +124,7 @@ struct acp_dev_data { struct snd_soc_dai_driver *dai_driver; int num_dai; - struct acp_stream *stream[ACP_MAX_STREAM]; + struct list_head stream_list; struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; -- 2.25.1

4 3

[PATCH v2 0/4] Add support for CLSA0101
by Lucas Tanure 28 Jul '22

28 Jul '22

Add Support for the CLSA0101 laptop, an Intel version of CLSA0100. This patch has been tested using the CLSA0100, ensuring it doesn't break the sound for it. We appreciate it if someone with CLSA0101 could verify that this the patch works for them. Changes from V1: - Add CLSA0101 id into scan.c, serial-multi-instantiate.c and cs35l41_hda_i2c.c Lucas Tanure (4): ALSA: hda: cs35l41: Use the CS35L41 HDA internal define ALSA: hda: cs35l41: Support CLSA0101 ACPI: scan: Add CLSA0101 Laptop Support platform/x86: serial-multi-instantiate: Add CLSA0101 Laptop drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 1 + sound/pci/hda/cs35l41_hda.c | 67 ++++++++++++------- sound/pci/hda/cs35l41_hda_i2c.c | 3 + sound/pci/hda/patch_realtek.c | 12 ++++ 5 files changed, 58 insertions(+), 26 deletions(-) -- 2.37.1

3 6

Add correct support for Rane SL-1
by GitHub pull_request - edited 28 Jul '22

28 Jul '22

alsa-project/alsa-ucm-conf pull request #187 was edited from christophehenry: Following [this ticket](https://bugzilla.kernel.org/show_bug.cgi?id=216082) on kernel bugzilla, I'm trying to add corect support for [Rane SL-1](https://cdn.korn.eu/pictures/product/400/040506_3.jpg), an old USB soundcard featuring 2 stereo output (RCA) and 3 stereo input (2 RCA, 1 6.3mm jack). Currently, `alsa-info.sh` does show the correct number of inputs and outputs But neither `pavucontrol` nor Mixxx do. I was told complex hardware should be described via UCM but the syntax is unfortunately very poorly documented so I can get my head around it and get `pavucontrol` to show all the inputs and outputs. Here's what I have so far which, to my disappointment, does not work. If I put only 1 `SectionDevice` section in the file, I can display 1 ouput under `pavucontrol`. With 2 `SectionDevice`, nothing shows. Without any UCM file, `pavucontrol` shows 2 outputs, 1 intputs (instead of 3) and Mixxx shows 2 outputs and not input. [The result of `alsa-info.sh` is available here](https://bugzilla.kernel.org/attachment.cgi?id=301351). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/187 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/187.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Biju Das 28 Jul '22

28 Jul '22

We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0d0594a0e4f6..c08ef317e3e3 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_dma; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_dma: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.25.1

1 1

[PATCH v3 0/2] ASoC: atmel: one fix and one cleanup
by Claudiu Beznea 28 Jul '22

28 Jul '22

Hi, The series adds one fix for mchp-spdifrx and one cleanups for mchp-spdifrx and mchp-spdifrx drivers. Thank you, Claudiu Beznea Changes in v3: - changed cover letter title s/few/one, s/cleanups/cleanup - fix compilation error and warnings - keep only patch 1/5 and patch 3/5 from previous version as the rest of them were integrated Changes in v2: - s/tag/tab in the title of patch 2/5 Claudiu Beznea (2): ASoC: mchp-spdifrx: disable end of block interrupt on failures ASoC: mchp-spdiftx: remove references to mchp_i2s_caps sound/soc/atmel/mchp-spdifrx.c | 9 ++++++--- sound/soc/atmel/mchp-spdiftx.c | 8 -------- 2 files changed, 6 insertions(+), 11 deletions(-) -- 2.34.1

2 3

[PATCH] ASoC: dt-bindings: use spi-peripheral-props.yaml
by Krzysztof Kozlowski 28 Jul '22

28 Jul '22

Instead of listing directly properties typical for SPI peripherals, reference the spi-peripheral-props.yaml schema. This allows using all properties typical for SPI-connected devices, even these which device bindings author did not tried yet. Remove the spi-* properties which now come via spi-peripheral-props.yaml schema, except for the cases when device schema adds some constraints like maximum frequency. While changing additionalProperties->unevaluatedProperties, put it in typical place, just before example DTS. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Technically, this depends on [1] merged to SPI tree, if we want to preserve existing behavior of not allowing SPI CPHA and CPOL in each of schemas in this patch. If this patch comes independently via different tree, the SPI CPHA and CPOL will be allowed for brief period of time, before [1] is merged. This will not have negative impact, just DT schema checks will be loosened for that period. [1] https://lore.kernel.org/all/20220722191539.90641-2-krzysztof.kozlowski@lina… --- .../devicetree/bindings/sound/adi,adau1977.yaml | 7 ++++--- Documentation/devicetree/bindings/sound/wlf,wm8731.yaml | 9 +++++---- 2 files changed, 9 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml index b80454ad97da..847b83398d3d 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml @@ -32,8 +32,6 @@ properties: reset-gpios: maxItems: 1 - spi-max-frequency: true - AVDD-supply: description: Analog power support for the device. @@ -52,7 +50,10 @@ required: - compatible - AVDD-supply -additionalProperties: false +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml index e7220e8b49f0..15795f63b5a3 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8731.yaml @@ -52,10 +52,6 @@ properties: DCVDD-supply: description: Digital core supply regulator for the DCVDD pin. - spi-max-frequency: true - -additionalProperties: false - required: - reg - compatible @@ -64,6 +60,11 @@ required: - DBVDD-supply - DCVDD-supply +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +unevaluatedProperties: false + examples: - | spi { -- 2.34.1

3 2

[PATCH] ASoC: codecs: va-macro: use fsgen as clock
by Srinivas Kandagatla 27 Jul '22

27 Jul '22

VA Macro fsgen clock is supplied to other LPASS Macros using proper clock apis, however the internal user uses the registers directly without clk apis. This approch has race condition where in external users of the clock might cut the clock while VA macro is actively using this. Moving the internal usage to clk apis would provide a proper refcounting and avoid such race conditions. This issue was noticed while headset was pulled out while recording is in progress and shifting record patch to DMIC. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index d18b56e60433..1ea10dc70748 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -199,6 +199,7 @@ struct va_macro { struct clk *mclk; struct clk *macro; struct clk *dcodec; + struct clk *fsgen; struct clk_hw hw; struct lpass_macro *pds; @@ -467,9 +468,9 @@ static int va_macro_mclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - return va_macro_mclk_enable(va, true); + return clk_prepare_enable(va->fsgen); case SND_SOC_DAPM_POST_PMD: - return va_macro_mclk_enable(va, false); + clk_disable_unprepare(va->fsgen); } return 0; @@ -1473,6 +1474,12 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_clkout; + va->fsgen = clk_hw_get_clk(&va->hw, "fsgen"); + if (IS_ERR(va->fsgen)) { + ret = PTR_ERR(va->fsgen); + goto err_clkout; + } + ret = devm_snd_soc_register_component(dev, &va_macro_component_drv, va_macro_dais, ARRAY_SIZE(va_macro_dais)); -- 2.25.1

3 2

snd_soc_skl - No sound on PopOS
by GitHub issues - opened 27 Jul '22

27 Jul '22

alsa-project/alsa-lib issue #255 was opened from runcros: Hello, I have no sound on skylake nau8825 max98357a here is my alsa-info : http://alsa-project.org/db/?f=c1b90f94c1bb072f507636f3979c0bc8a62ae470 Can you help me please If you need others infos, I am here =) thank you ! Issue URL : https://github.com/alsa-project/alsa-lib/issues/255 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: mediatek: mt8186: remove unnecessary judgments
by Jiaxin Yu 27 Jul '22

27 Jul '22

The afe_priv->dai_priv[] is allocated when platform driver probe(), if it failed, the ASoC platform driver probe() will return fail first. Therefore, this is excessive judgment, and the condition will never be established. Bug report: https://www.spinics.net/lists/alsa-devel/msg145609.html This is a semi-automatic email about new static checker warnings. The patch ae92dcbee8b6: "ASoC: mediatek: mt8186: support tdm in platform driver" from May 23, 2022, leads to the following Smatch complaint: sound/soc/mediatek/mt8186/mt8186-dai-tdm.c:424 mtk_dai_tdm_hw_params() warn: variable dereferenced before check 'tdm_priv' (see line 406) sound/soc/mediatek/mt8186/mt8186-dai-tdm.c 405 struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[tdm_id]; 406 unsigned int tdm_mode = tdm_priv->tdm_mode; ^^^^^^^^^^^^^^^^^^^ Lot's of dereferences 407 unsigned int data_mode = tdm_priv->data_mode; 408 unsigned int rate = params_rate(params); 409 unsigned int channels = params_channels(params); 410 snd_pcm_format_t format = params_format(params); 411 unsigned int bit_width = 412 snd_pcm_format_physical_width(format); 413 unsigned int tdm_channels = (data_mode == TDM_DATA_ONE_PIN) ? 414 get_tdm_ch_per_sdata(tdm_mode, channels) : 2; 415 unsigned int lrck_width = 416 get_tdm_lrck_width(format, tdm_mode); 417 unsigned int tdm_con = 0; 418 bool slave_mode = tdm_priv->slave_mode; 419 bool lrck_inv = tdm_priv->lck_invert; 420 bool bck_inv = tdm_priv->bck_invert; 421 unsigned int tran_rate; 422 unsigned int tran_relatch_rate; 423 424 if (!tdm_priv) { ^^^^^^^^^ Checked too late 425 dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); 426 return -EINVAL; Fixes: ae92dcbee8b6 ("ASoC: mediatek: mt8186: support tdm in platform driver") Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 5 -- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 63 --------------------- sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 5 -- sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 55 +----------------- 4 files changed, 1 insertion(+), 127 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index 6be6d4f3b585..f6f23c7c3ff5 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -655,11 +655,6 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", __func__, id, substream->stream, rate); - if (!adda_priv) { - dev_err(afe->dev, "%s(), adda_priv == NULL", __func__); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { unsigned int dl_src2_con0; unsigned int dl_src2_con1; diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c index 5c1290b950e8..ec79e2f2a54d 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -119,12 +119,6 @@ static int mt8186_i2s_hd_get(struct snd_kcontrol *kcontrol, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - ucontrol->value.integer.value[0] = i2s_priv->low_jitter_en; return 0; @@ -148,12 +142,6 @@ static int mt8186_i2s_hd_set(struct snd_kcontrol *kcontrol, __func__, kcontrol->id.name, hd_en); i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - if (i2s_priv->low_jitter_en == hd_en) return 0; @@ -377,11 +365,6 @@ static int mtk_i2s_en_event(struct snd_soc_dapm_widget *w, i2s_priv = get_i2s_priv_by_name(afe, w->name); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", __func__, w->name, event); @@ -442,11 +425,6 @@ static int mtk_mclk_en_event(struct snd_soc_dapm_widget *w, i2s_priv = get_i2s_priv_by_name(afe, w->name); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - switch (event) { case SND_SOC_DAPM_PRE_PMU: mt8186_mck_enable(afe, i2s_priv->mclk_id, i2s_priv->mclk_rate); @@ -566,12 +544,6 @@ static int mtk_afe_i2s_share_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (i2s_priv->share_i2s_id < 0) return 0; @@ -587,12 +559,6 @@ static int mtk_afe_i2s_hd_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (get_i2s_id_by_name(afe, sink->name) == get_i2s_id_by_name(afe, source->name)) return i2s_priv->low_jitter_en; @@ -618,15 +584,8 @@ static int mtk_afe_i2s_apll_connect(struct snd_soc_dapm_widget *source, int i2s_need_apll; i2s_priv = get_i2s_priv_by_name(afe, w->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); - /* choose APLL from i2s rate */ i2s_need_apll = mt8186_get_apll_by_rate(afe, i2s_priv->rate); @@ -642,12 +601,6 @@ static int mtk_afe_i2s_mclk_connect(struct snd_soc_dapm_widget *source, struct mtk_afe_i2s_priv *i2s_priv; i2s_priv = get_i2s_priv_by_name(afe, sink->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - if (get_i2s_id_by_name(afe, sink->name) == get_i2s_id_by_name(afe, source->name)) return (i2s_priv->mclk_rate > 0) ? 1 : 0; @@ -672,12 +625,6 @@ static int mtk_afe_mclk_apll_connect(struct snd_soc_dapm_widget *source, int cur_apll; i2s_priv = get_i2s_priv_by_name(afe, w->name); - - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); @@ -980,11 +927,6 @@ static int mtk_dai_i2s_config(struct mtk_base_afe *afe, dev_dbg(afe->dev, "%s(), id %d, rate %d, format %d\n", __func__, i2s_id, rate, format); - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - i2s_priv->rate = rate; switch (i2s_id) { @@ -1053,11 +995,6 @@ static int mtk_dai_i2s_set_sysclk(struct snd_soc_dai *dai, int apll; int apll_rate; - if (!i2s_priv) { - dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); - return -EINVAL; - } - if (dir != SND_SOC_CLOCK_OUT) { dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); return -EINVAL; diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c index 0b0032ecfe6d..41221a66111c 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c @@ -287,11 +287,6 @@ static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_pcm_priv *pcm_priv = afe_priv->dai_priv[dai->id]; - if (!pcm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - /* DAI mode*/ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c index c6ead7c252f0..4148dceb3a4c 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c @@ -106,11 +106,6 @@ static int mtk_tdm_en_event(struct snd_soc_dapm_widget *w, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", __func__, w->name, event); @@ -138,11 +133,6 @@ static int mtk_tdm_mck_en_event(struct snd_soc_dapm_widget *w, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x, dai_id %d\n", __func__, w->name, event, dai_id); @@ -215,11 +205,6 @@ static int mtk_afe_tdm_mclk_connect(struct snd_soc_dapm_widget *source, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - return (tdm_priv->mclk_rate > 0) ? 1 : 0; } @@ -250,11 +235,6 @@ static int mtk_afe_tdm_hd_connect(struct snd_soc_dapm_widget *source, int dai_id = get_tdm_id_by_name(w->name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - return tdm_priv->low_jitter_en; } @@ -270,11 +250,6 @@ static int mtk_afe_tdm_apll_connect(struct snd_soc_dapm_widget *source, int cur_apll; int tdm_need_apll; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return 0; - } - /* which apll */ cur_apll = mt8186_get_apll_by_name(afe, source->name); @@ -303,11 +278,6 @@ static int mt8186_tdm_hd_get(struct snd_kcontrol *kcontrol, int dai_id = get_tdm_id_by_name(kcontrol->id.name); struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - ucontrol->value.integer.value[0] = tdm_priv->low_jitter_en; return 0; @@ -332,11 +302,6 @@ static int mt8186_tdm_hd_set(struct snd_kcontrol *kcontrol, dev_dbg(afe->dev, "%s(), kcontrol name %s, hd_en %d\n", __func__, kcontrol->id.name, hd_en); - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - if (tdm_priv->low_jitter_en == hd_en) return 0; @@ -421,22 +386,14 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, unsigned int tran_rate; unsigned int tran_relatch_rate; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - tdm_priv->rate = rate; - tran_rate = mt8186_rate_transform(afe->dev, rate, dai->id); tran_relatch_rate = mt8186_tdm_relatch_rate_transform(afe->dev, rate); /* calculate mclk_rate, if not set explicitly */ if (!tdm_priv->mclk_rate) { tdm_priv->mclk_rate = rate * tdm_priv->mclk_multiple; - mtk_dai_tdm_cal_mclk(afe, - tdm_priv, - tdm_priv->mclk_rate); + mtk_dai_tdm_cal_mclk(afe, tdm_priv, tdm_priv->mclk_rate); } /* ETDM_IN1_CON0 */ @@ -508,11 +465,6 @@ static int mtk_dai_tdm_set_sysclk(struct snd_soc_dai *dai, struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - if (dir != SND_SOC_CLOCK_IN) { dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); return -EINVAL; @@ -529,11 +481,6 @@ static int mtk_dai_tdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct mt8186_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; - if (!tdm_priv) { - dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); - return -EINVAL; - } - /* DAI mode*/ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: -- 2.25.1

2 1

[PATCH] ASoC: amd: acp: uninitialized variable in i2s_irq_handler()
by Dan Carpenter 27 Jul '22

27 Jul '22

The "ext_intr_stat1" is used without being initialized. Fixes: 93f53881473c ("ASoC: amd: acp: Modify local variables name to generic") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/acp/acp-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 10730d33c3b0..f561d39b33e2 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -100,7 +100,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) return IRQ_NONE; if (adata->rsrc->no_of_ctrls == 2) - ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); + ext_intr_stat1 = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); -- 2.35.1

2 1

[PATCH 0/2] Add support for CLSA0101
by Lucas Tanure 27 Jul '22

27 Jul '22

Add Support for the CLSA0101 laptop, an Intel version of CLSA0100. This patch has been tested using the CLSA0100, ensuring it doesn't break the sound for it. We appreciate it if someone with CLSA0101 could verify that this the patch works for them. Lucas Tanure (2): ALSA: hda: cs35l41: Use the CS35L41 HDA internal define ALSA: hda: cs35l41: Support CLSA0101 sound/pci/hda/cs35l41_hda.c | 67 +++++++++++++++++++++-------------- sound/pci/hda/patch_realtek.c | 12 +++++++ 2 files changed, 53 insertions(+), 26 deletions(-) -- 2.37.1

3 6

[PATCH 0/4] ALSA: Defer async signal handling
by Takashi Iwai 27 Jul '22

27 Jul '22

Hi, this series is another attempt to address the potential deadlocks via kill_fasync() calls that have been reported by syzbot for long time. Instead of the previous series to drop the async handler[*], this tries to defer the kill_fasync() call. A few new common helpers are introduced at first, then the actual usages are replaced in the following patches. The patches passed the quick tests with alsa-lib test cases. Takashi [*] https://lore.kernel.org/r/20220717070549.5993-1-tiwai@suse.de === Takashi Iwai (4): ALSA: core: Add async signal helpers ALSA: timer: Use deferred fasync helper ALSA: pcm: Use deferred fasync helper ALSA: control: Use deferred fasync helper include/sound/control.h | 2 +- include/sound/core.h | 8 ++++ include/sound/pcm.h | 2 +- sound/core/control.c | 7 ++-- sound/core/misc.c | 93 +++++++++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/timer.c | 11 ++--- 9 files changed, 116 insertions(+), 12 deletions(-) -- 2.35.3

2 6

HDA ATI HDMI not supported
by GitHub issues - opened 27 Jul '22

27 Jul '22

alsa-project/alsa-ucm-conf issue #194 was opened from yangxiaohua2009: I am using the latest ucm2 file and alsalib alsainfo [http://alsa-project.org/db/?f=92c58f3a6b72fee05b678386858dbc816d25dbf3](htt… When nothing is changed, the alsaucm reload command returns error saying: `ALSA lib parser.c:244:(error_node) UCM is not supported for this HDA model (HDA ATI HDMI at 0xfeb60000 irq 28)` and the HDMI is not working. After changing the file /ucm2/HDA/HDA.conf, from `Define.AcpCardId "$${find-card:field=name,return=id,regex='^(acp|acp6x)$'}"` to `Define.AcpCardId "$${find-card:field=name,return=id,regex='^(acp|acp6x|acpes8336)$'}"` alsucm reload returns no error, but alsaucm listcards returns: `ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:1 use case configuration -2` ` 0: hw:0` ` HDA ATI HDMI at 0xfeb60000 irq 28` despite the fact that I already added the acpes8336.conf in conf.d and acpes8336 folder. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/194 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: mediatek: mt8186: set the correct string to strncmp()
by Jiaxin Yu 26 Jul '22

26 Jul '22

Fix Smatch static checker warning. strncmp() here only needs to compare the first seven bytes, so in order to make the code more clear, only the first seven bytes of the string used as the comparison are reserved. Bug report: https://www.spinics.net/lists/alsa-devel/msg145608.html sound/soc/mediatek/mt8186/mt8186-dai-adda.c:78 get_adda_priv_by_name() warn: strncmp() with weird length: 17 vs 7 sound/soc/mediatek/mt8186/mt8186-dai-adda.c 72 static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, 73 const char *name) 74 { 75 struct mt8186_afe_private *afe_priv = afe->platform_priv; 76 int dai_id; 77 --> 78 if (strncmp(name, "aud_dac_hires_clk", 7) == 0 || 79 strncmp(name, "aud_adc_hires_clk", 7) == 0) Fixes: b65c466220b3 ("ASoC: mediatek: mt8186: support adda in platform driver") Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index db71b032770d..81976f7c444e 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -75,8 +75,7 @@ static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, struct mt8186_afe_private *afe_priv = afe->platform_priv; int dai_id; - if (strncmp(name, "aud_dac_hires_clk", 7) == 0 || - strncmp(name, "aud_adc_hires_clk", 7) == 0) + if (strncmp(name, "aud_dac", 7) == 0 || strncmp(name, "aud_adc", 7) == 0) dai_id = MT8186_DAI_ADDA; else return NULL; -- 2.25.1

2 1

[PATCH v1] ASoC: Intel: cirrus-common: Use UID to map correct amp to prefix
by Stefan Binding 26 Jul '22

26 Jul '22

Since the order of the amps in the ACPI determines the device name, and the ACPI order may change depending on hardware configuration, use UID to dynamically compute the dai links, allowing dynamic assignment of the name_prefix. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 73 +++++++++------------- 1 file changed, 31 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index f4192df962d6..a96bc70a34d7 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -10,6 +10,9 @@ #include "../../codecs/cs35l41.h" #include "sof_cirrus_common.h" +#define CS35L41_HID "CSC3541" +#define CS35L41_MAX_AMPS 4 + /* * Cirrus Logic CS35L41/CS35L53 */ @@ -35,50 +38,12 @@ static const struct snd_soc_dapm_route cs35l41_dapm_routes[] = { {"TR Spk", NULL, "TR SPK"}, }; -static struct snd_soc_dai_link_component cs35l41_components[] = { - { - .name = CS35L41_DEV0_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV1_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV2_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV3_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, -}; +static struct snd_soc_dai_link_component cs35l41_components[CS35L41_MAX_AMPS]; /* * Mapping between ACPI instance id and speaker position. - * - * Four speakers: - * 0: Tweeter left, 1: Woofer left - * 2: Tweeter right, 3: Woofer right */ -static struct snd_soc_codec_conf cs35l41_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TR", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "WR", - }, -}; +static struct snd_soc_codec_conf cs35l41_codec_conf[CS35L41_MAX_AMPS]; static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) { @@ -117,10 +82,10 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) static const struct { unsigned int rx[2]; } cs35l41_channel_map[] = { - {.rx = {0, 1}}, /* TL */ {.rx = {0, 1}}, /* WL */ - {.rx = {1, 0}}, /* TR */ {.rx = {1, 0}}, /* WR */ + {.rx = {0, 1}}, /* TL */ + {.rx = {1, 0}}, /* TR */ }; static int cs35l41_hw_params(struct snd_pcm_substream *substream, @@ -175,8 +140,32 @@ static const struct snd_soc_ops cs35l41_ops = { .hw_params = cs35l41_hw_params, }; +static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; + +static const char * const cs35l41_uid_strings[] = { "0", "1", "2", "3" }; + +static void cs35l41_compute_codec_conf(void) +{ + int uid; + struct acpi_device *adev; + struct device *physdev; + + for (uid = 0; uid < CS35L41_MAX_AMPS; uid++) { + adev = acpi_dev_get_first_match_dev(CS35L41_HID, cs35l41_uid_strings[uid], -1); + if (!adev) + return; + physdev = get_device(acpi_get_first_physical_node(adev)); + cs35l41_components[uid].name = dev_name(physdev); + cs35l41_components[uid].dai_name = CS35L41_CODEC_DAI; + cs35l41_codec_conf[uid].dlc.name = dev_name(physdev); + cs35l41_codec_conf[uid].name_prefix = cs35l41_name_prefixes[uid]; + acpi_dev_put(adev); + } +} + void cs35l41_set_dai_link(struct snd_soc_dai_link *link) { + cs35l41_compute_codec_conf(); link->codecs = cs35l41_components; link->num_codecs = ARRAY_SIZE(cs35l41_components); link->init = cs35l41_init; -- 2.34.1

3 2

[PATCH v7] ASoc: dt-bindings: tas2780: remove tas2780 and will merge it to tas27xx
by Raphael-Xu 26 Jul '22

26 Jul '22

remove tas2780 Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../devicetree/bindings/sound/tas2780.yaml | 75 ------------------- 1 file changed, 75 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2780.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2780.yaml b/Documentation/devicetree/bindings/sound/tas2780.yaml deleted file mode 100644 index 21389a16fdb9..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2780.yaml +++ /dev/null @@ -1,75 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -# Copyright (C) 2020-2022 Texas Instruments Incorporated -%YAML 1.2 ---- -$id: "http://devicetree.org/schemas/sound/tas2780.yaml#" -$schema: "http://devicetree.org/meta-schemas/core.yaml#" - -title: Texas Instruments TAS2780 Smart PA - -maintainers: - - Raphael Xu <raphael-xu(a)ti.com> - -description: | - The TAS2780 is a mono, digital input Class-D audio amplifier optimized for - efficiently driving high peak power into small loudspeakers. - Integrated speaker voltage and current sense provides for - real time monitoring of loudspeaker behavior. - -properties: - compatible: - enum: - - ti,tas2780 - - reg: - description: - I2C address of the device can be between 0x38 to 0x3f. - minimum: 0x38 - maximum: 0x3f - - reset-gpios: - description: GPIO used to reset the device. - maxItems: 1 - - interrupts: - maxItems: 1 - - ti,imon-slot-no: - description: TDM TX current sense time slot. - $ref: /schemas/types.yaml#/definitions/uint32 - minimum: 0 - maximum: 3 - - ti,vmon-slot-no: - description: TDM TX voltage sense time slot. - $ref: /schemas/types.yaml#/definitions/uint32 - minimum: 0 - maximum: 3 - - '#sound-dai-cells': - const: 1 - -required: - - compatible - - reg - -additionalProperties: false - -examples: - - | - #include <dt-bindings/gpio/gpio.h> - i2c0 { - #address-cells = <1>; - #size-cells = <0>; - codec: codec@38 { - compatible = "ti,tas2780"; - reg = <0x38>; - #sound-dai-cells = <1>; - interrupts = <14>; - reset-gpios = <&gpio1 15 0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <2>; - }; - }; - -... -- 2.35.1

3 4

rt715-sdca: use sensible capture gain value
by GitHub pull_request - opened 26 Jul '22

26 Jul '22

alsa-project/alsa-ucm-conf pull request #193 was opened from plbossart: The value of 124 used for FU02 is way too much, with saturation even in low-volume cases. The 0dB value (47) is a much better initialization value. BugLink: https://github.com/thesofproject/linux/issues/3766 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/193 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/193.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: dt-bindings: qcom, wcd934x: use absolute path to other schema
by Krzysztof Kozlowski 26 Jul '22

26 Jul '22

Absolute path to other DT schema is preferred over relative one. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml index 9b225dbf8b79..8ca19f2b0b3d 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -127,7 +127,7 @@ properties: gpio@42: type: object - $ref: ../gpio/qcom,wcd934x-gpio.yaml# + $ref: /schemas/gpio/qcom,wcd934x-gpio.yaml# patternProperties: "^.*@[0-9a-f]+$": -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: Fix initialization of ext_intr_stat1 in i2s_irq_handler()
by Nathan Chancellor 26 Jul '22

26 Jul '22

Clang warns: ../sound/soc/amd/acp/acp-platform.c:117:19: error: variable 'ext_intr_stat1' is uninitialized when used here [-Werror,-Wuninitialized] if (stream && (ext_intr_stat1 & stream->irq_bit)) { ^~~~~~~~~~~~~~ ../sound/soc/amd/acp/acp-platform.c:97:35: note: initialize the variable 'ext_intr_stat1' to silence this warning u32 ext_intr_stat, ext_intr_stat1, i; ^ = 0 1 error generated. The variable was not properly renamed, correct it to resolve the warning. Fixes: 93f53881473c ("ASoC: amd: acp: Modify local variables name to generic") Link: https://github.com/ClangBuiltLinux/linux/issues/1675 Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/amd/acp/acp-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 10730d33c3b0..f561d39b33e2 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -100,7 +100,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) return IRQ_NONE; if (adata->rsrc->no_of_ctrls == 2) - ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); + ext_intr_stat1 = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); base-commit: be0aa8d4b0fcb4532bf7973141e911998ab39508 -- 2.37.1

3 2

[PATCH v2 0/5] ASoC: atmel: one fix and few cleanups
by Claudiu Beznea 26 Jul '22

26 Jul '22

Hi, The series adds one fix for mchp-spdifrx and few cleanups for mchp-spdifrx and mchp-spdifrx drivers. Thank you, Claudiu Beznea Changes in v2: - s/tag/tab in the title of patch 2/5 Claudiu Beznea (5): ASoC: mchp-spdifrx: disable end of block interrupt on failures ASoC: mchp-spdifrx: use single tag indent for structure ASoC: mchp-spdiftx: remove references to mchp_i2s_caps ASoC: mchp-spdiftx: return directly ret ASoC: mchp-spdiftx: add and remove black line around MODULE_DEVICE_TABLE() sound/soc/atmel/mchp-spdifrx.c | 19 +++++++++++-------- sound/soc/atmel/mchp-spdiftx.c | 21 +++++---------------- 2 files changed, 16 insertions(+), 24 deletions(-) -- 2.34.1

2 8

[PATCH 0/5] ASoC: atmel: one fix and few cleanups
by Claudiu Beznea 26 Jul '22

26 Jul '22

Hi, The series adds one fix for mchp-spdifrx and few cleanups for mchp-spdifrx and mchp-spdifrx drivers. Thank you, Claudiu Beznea Claudiu Beznea (5): ASoC: mchp-spdifrx: disable end of block interrupt on failures ASoC: mchp-spdifrx: use single tag indent for structure ASoC: mchp-spdiftx: remove references to mchp_i2s_caps ASoC: mchp-spdiftx: return directly ret ASoC: mchp-spdiftx: add and remove black line around MODULE_DEVICE_TABLE() sound/soc/atmel/mchp-spdifrx.c | 19 +++++++++++-------- sound/soc/atmel/mchp-spdiftx.c | 21 +++++---------------- 2 files changed, 16 insertions(+), 24 deletions(-) -- 2.34.1

2 6

[PATCH v2 1/2] ASoC: codecs: add support for ES8326
by Zhu Ning 26 Jul '22

26 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ----- v2 changes since v1: 1. add i2c_id_table 2. add .endianness .use_pmdown_time flag 3. fix dt-binding name --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 809 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 +++++++++ 4 files changed, 1003 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..d387f47c4bf4 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,809 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS_1B, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS_1B, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DETECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + es8326_disable_micbias(comp); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if (!es8326->jack) + goto out; + + es8326_enable_micbias(comp); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8326->regmap, false); + regcache_sync(es8326->regmap); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + es8326_disable_micbias(component); + cancel_delayed_work_sync(&es8326->jack_detect_work); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_OFF); + regcache_cache_only(es8326->regmap, true); + regcache_mark_dirty(es8326->regmap); + + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) { + es8326_disable_micbias(component); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + } + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) { + dev_warn(&i2c->dev, "Failed to reguest IRQ: %d: %d\n", + es8326->irq, ret); + es8326->irq = -ENXIO; + } + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, + .id_table = es8326_i2c_id, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..717f3fd42372 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.h -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

3 4

[PATCH 0/3] ASoC: SOF: minor updates for 5.20
by Pierre-Louis Bossart 26 Jul '22

26 Jul '22

One sanity check for SSP index reported by NHLT/BIOS and two updates for Mediatek and Intel Chromebooks related to already-merged firmware changes. Ideally this should go in 5.20, time-permitting. Brent Lu (1): ASoC: SOF: dai-intel: add SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_AON bit Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda: add sanity check on SSP index reported by NHLT YC Hung (1): ASoC: SOF: Add cont_update_posn to platform parameters include/sound/sof/dai-intel.h | 2 ++ include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/intel/hda.c | 7 +++++++ sound/soc/sof/ipc3-pcm.c | 3 +++ sound/soc/sof/sof-priv.h | 1 + 5 files changed, 14 insertions(+), 1 deletion(-) -- 2.34.1

2 4

[PATCH 00/10] ASoC: Intel: machine driver updates for 5.20
by Pierre-Louis Bossart 26 Jul '22

26 Jul '22

A couple of minor updates contributed and tested by Intel teams or end-users. Ideally these should be applied for 5.20, time-permitting. Andrey Turkin (2): ASoC: Intel: sof_es8336: Fix GPIO quirks set via module option ASoC: Intel: sof_es8336: ignore GpioInt when looking for speaker/headset GPIO lines Muralidhar Reddy (3): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in ADL match table ASoC: Intel: sof_es8336: add support for HDMI_In capture ASoC: Intel: sof_es8336: reset the num_links during probe Pierre-Louis Bossart (2): ASoC: Intel: sof_sdw: add support for Dell SKU 0AF0 ASoC: Intel: sof_es8336: remove hard-coded SSP selection Yong Zhi (3): ASoC: Intel: sof_rt5682: Add support for mtl_mx98357_rt5682 ASoC: Intel: sof_rt5682: Perform quirk check first in card late probe ASoC: Intel: sof_nau8825: Move quirk check to the front in late probe sound/hda/intel-dsp-config.c | 5 + sound/soc/intel/boards/sof_es8336.c | 160 +++++++++++++----- sound/soc/intel/boards/sof_nau8825.c | 10 +- sound/soc/intel/boards/sof_rt5682.c | 26 ++- sound/soc/intel/boards/sof_sdw.c | 10 ++ .../intel/common/soc-acpi-intel-adl-match.c | 25 +++ .../intel/common/soc-acpi-intel-mtl-match.c | 17 ++ 7 files changed, 201 insertions(+), 52 deletions(-) -- 2.34.1

2 11

[PATCH -next] ASoC: mediatek: mt8186: fix Kconfig dependency
by Ren Zhijie 26 Jul '22

26 Jul '22

If CONFIG_SND_SOC_MT8186=y and CONFIG_SND_SOC_MT6358 is not set, make ARCH=x86_64 CROSS_COMPILE=x86_64-linux-gnu-, will be failed, like this: sound/soc/mediatek/mt8186/mt8186-mt6366-common.o: In function `mt8186_mt6366_init': mt8186-mt6366-common.c:(.text+0x4d): undefined reference to `mt6358_set_mtkaif_protocol' make: *** [vmlinux] Error 1 To fix this build error, add select SND_SOC_MT6358 to config SND_SOC_MT8186 dependency. Fixes: 097e874ad3fc ("ASoC: mediatek: mt8186: add platform driver") Signed-off-by: Ren Zhijie <renzhijie2(a)huawei.com> --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 96917743b0db..363fa4d47680 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -157,6 +157,7 @@ config SND_SOC_MT8186 depends on ARCH_MEDIATEK || COMPILE_TEST depends on COMMON_CLK select SND_SOC_MEDIATEK + select SND_SOC_MT6358 select MFD_SYSCON if SND_SOC_MT6358 help This adds ASoC driver for Mediatek MT8186 boards -- 2.17.1

2 1

[PATCH] ASoC: max98373: Removing 0x203E from the volatile reg
by Ryan Lee 26 Jul '22

26 Jul '22

The 0x203E speaker gain register should be non-volatile. This register value was not able to be synced because it was marked as volatile. Signed-off-by: Ryan Lee <ryans.lee(a)analog.com> --- sound/soc/codecs/max98373-i2c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 4fe065ece17c..3e04c7f0cce4 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -442,7 +442,6 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98373_R2000_SW_RESET ... MAX98373_R2009_INT_FLAG3: - case MAX98373_R203E_AMP_PATH_GAIN: case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: -- 2.37.0

3 2

[PATCH] ASoC: ti: remove sound support for DaVinci DM644x and DM646x
by Lukas Bulwahn 26 Jul '22

26 Jul '22

Commit 7dd33764486d ("ARM: davinci: Delete DM644x board files") and commit b4aed01de486 ("ARM: davinci: Delete DM646x board files") removes the support for DaVinci DM644x and DM646x boards. Hence, remove the sound support for those boards as well. Signed-off-by: Lukas Bulwahn <lukas.bulwahn(a)gmail.com> --- sound/soc/ti/Kconfig | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 40110e9a9e8a..35fddd7c501a 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -180,9 +180,6 @@ config SND_SOC_DAVINCI_EVM select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_DM355_EVM select SND_SOC_DAVINCI_ASP if SND_SOC_DM365_AIC3X_CODEC select SND_SOC_DAVINCI_VCIF if SND_SOC_DM365_VOICE_CODEC - select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_EVM # DM6446 - select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DM6467_EVM - select SND_SOC_SPDIF if MACH_DAVINCI_DM6467_EVM select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA830_EVM select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA850_EVM select SND_SOC_TLV320AIC3X @@ -191,8 +188,6 @@ config SND_SOC_DAVINCI_EVM DaVinci EVM platforms: - DM355 - DM365 - - DM6446 - - DM6447 - DM830 - DM850 -- 2.17.1

3 2

Question about DPCM locking
by Kuninori Morimoto 26 Jul '22

26 Jul '22

Hi Takashi May I ask about this patch ? ASoC: soc-pcm: Fix and cleanup DPCM locking b7898396f4bbe160f546d0c5e9fa17cca9a7d153 Q1. It exchanges many function parameters - func(substream, xxx) + func(rtd, substream, xxx) ^^^^ I guess the purpose is it want to call snd_soc_dpcm_mutex_assert_held() inside the func ? But it looks very verbose. we can do like this, but am I misunderstanding ? func(substream, xxx) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_assert_held(rtd); ... } Q2. It added new __soc_pcm_close(). But soc_pcm_close() is using soc_pcm_clean() instead of __soc_pcm_close(). Is this just a mistake ? static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_lock(rtd); - soc_pcm_clean(substream, 0); + __soc_pcm_close(substream); snd_soc_dpcm_mutex_unlock(rtd); return 0; } Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

Re: [PATCH v8] ASoc: dt-bindings: tas27xx: revise tas27xx format
by Rob Herring 26 Jul '22

26 Jul '22

On Fri, Jul 22, 2022 at 06:55:12PM +0800, Raphael-Xu wrote: > revise the format Complete sentences. And this is not v8. Each new change is v1. > > Signed-off-by: Raphael-Xu <13691752556(a)139.com> > --- > .../devicetree/bindings/sound/tas27xx.yaml | 16 +++++++++++----- > 1 file changed, 11 insertions(+), 5 deletions(-) > > diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml > index 66a0df8850ea..573652577462 100644 > --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml > +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml > @@ -8,7 +8,7 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" > title: Texas Instruments TAS2764/TAS2780 Smart PA > > maintainers: > - - Shenghao Ding <shenghao-ding(a)ti.com> > + - Raphael Xu <raphael-xu(a)ti.com> > > description: | > The TAS2764/TAS2780 is a mono, digital input Class-D audio amplifier > @@ -23,17 +23,19 @@ properties: > - ti,tas2780 > > reg: > - maxItems: 1 > description: | > - I2C address of the device can be between 0x38 to 0x45. 0x40-0x45 is no longer valid? > + I2C address of the device can be between 0x38 to 0x3f. > + minimum: 0x38 > + maximum: 0x3f > > reset-gpios: > - maxItems: 1 > description: GPIO used to reset the device. > + maxItems: 1 Unnecessary change. > > shutdown-gpios: > + description: | Don't need '|' if no formatting. > + GPIO used to control the state of the device.Only for TAS2764. Space needed after period. > maxItems: 1 > - description: GPIO used to control the state of the device. > > interrupts: > maxItems: 1 > @@ -41,10 +43,14 @@ properties: > ti,imon-slot-no: > $ref: /schemas/types.yaml#/definitions/uint32 > description: TDM TX current sense time slot. > + minimum: 0 > + maximum: 3 > > ti,vmon-slot-no: > $ref: /schemas/types.yaml#/definitions/uint32 > description: TDM TX voltage sense time slot. > + minimum: 0 > + maximum: 3 > > '#sound-dai-cells': > const: 1 > -- > 2.35.1 > > >

1 0

[PATCH v2 0/5] ALSA: Drop async signal support
by Takashi Iwai 25 Jul '22

25 Jul '22

Hi, this is a revised patch set for dropping fasync support from ALSA core. The async signal itself is very difficult to use properly due to various restrictions (e.g. you cannot perform any I/O in the context), hence it's a feature that has been never used by real applications. OTOH, the real problem is that there have been quite a few syzcaller reports indicating that fasync code path may lead to some potential deadlocks for long time. Dropping the feature is the easiest solution, obviously. The corresponding update for alsa-lib will follow once when we agree with this approach. thanks, Takashi === v1: https://lore.kernel.org/r/20220715102935.4695-1-tiwai@suse.de v1->v2: Fix unused variable warning in patch 2 === Takashi Iwai (5): ALSA: timer: Drop async signal support ALSA: pcm: Drop async signal support ALSA: control: Drop async signal support ALSA: core: Drop async signal support ALSA: doc: Drop stale fasync entry .../kernel-api/writing-an-alsa-driver.rst | 1 - include/sound/control.h | 1 - include/sound/pcm.h | 1 - sound/core/control.c | 11 ----------- sound/core/init.c | 11 +---------- sound/core/pcm_lib.c | 8 +------- sound/core/pcm_native.c | 18 ------------------ sound/core/timer.c | 13 ------------- 8 files changed, 2 insertions(+), 62 deletions(-) -- 2.35.3

2 7

[bug report] ASoC: mediatek: mt8186: support tdm in platform driver
by Dan Carpenter 25 Jul '22

25 Jul '22

Hello Jiaxin Yu, This is a semi-automatic email about new static checker warnings. The patch ae92dcbee8b6: "ASoC: mediatek: mt8186: support tdm in platform driver" from May 23, 2022, leads to the following Smatch complaint: sound/soc/mediatek/mt8186/mt8186-dai-tdm.c:424 mtk_dai_tdm_hw_params() warn: variable dereferenced before check 'tdm_priv' (see line 406) sound/soc/mediatek/mt8186/mt8186-dai-tdm.c 405 struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[tdm_id]; 406 unsigned int tdm_mode = tdm_priv->tdm_mode; ^^^^^^^^^^^^^^^^^^^ Lot's of dereferences 407 unsigned int data_mode = tdm_priv->data_mode; 408 unsigned int rate = params_rate(params); 409 unsigned int channels = params_channels(params); 410 snd_pcm_format_t format = params_format(params); 411 unsigned int bit_width = 412 snd_pcm_format_physical_width(format); 413 unsigned int tdm_channels = (data_mode == TDM_DATA_ONE_PIN) ? 414 get_tdm_ch_per_sdata(tdm_mode, channels) : 2; 415 unsigned int lrck_width = 416 get_tdm_lrck_width(format, tdm_mode); 417 unsigned int tdm_con = 0; 418 bool slave_mode = tdm_priv->slave_mode; 419 bool lrck_inv = tdm_priv->lck_invert; 420 bool bck_inv = tdm_priv->bck_invert; 421 unsigned int tran_rate; 422 unsigned int tran_relatch_rate; 423 424 if (!tdm_priv) { ^^^^^^^^^ Checked too late 425 dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); 426 return -EINVAL; regards, dan carpenter

2 1

[bug report] ASoC: mediatek: mt8186: support adda in platform driver
by Dan Carpenter 25 Jul '22

25 Jul '22

Hello Jiaxin Yu, The patch b65c466220b3: "ASoC: mediatek: mt8186: support adda in platform driver" from May 23, 2022, leads to the following Smatch static checker warning: sound/soc/mediatek/mt8186/mt8186-dai-adda.c:78 get_adda_priv_by_name() warn: strncmp() with weird length: 17 vs 7 sound/soc/mediatek/mt8186/mt8186-dai-adda.c 72 static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, 73 const char *name) 74 { 75 struct mt8186_afe_private *afe_priv = afe->platform_priv; 76 int dai_id; 77 --> 78 if (strncmp(name, "aud_dac_hires_clk", 7) == 0 || 79 strncmp(name, "aud_adc_hires_clk", 7) == 0) 7 is weird. It would be more readable to say it as: if (strncmp(name, "aud_dac", 7) == 0 || strncmp(name, "aud_adc", 7) == 0) 80 dai_id = MT8186_DAI_ADDA; 81 else 82 return NULL; 83 84 return afe_priv->dai_priv[dai_id]; 85 } regards, dan carpenter

2 1

[PATCH] ALSA: hda/realtek: Enable speaker and mute LEDs for HP laptops
by Kai-Heng Feng 25 Jul '22

25 Jul '22

Two more HP laptops that use cs35l41 AMP for speaker and GPIO for mute LEDs. So use the existing quirk to enable them accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f55bc43bfa9c..cdbee71e83216 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9109,6 +9109,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), -- 2.36.1

3 5

[PATCH -next] ALSA: hda: cs35l41: Fix build error unused-function
by Ren Zhijie 25 Jul '22

25 Jul '22

If CONFIG_PM_SLEEP is not set, make ARCH=x86_64 CROSS_COMPILE=x86_64-linux-gnu-, will be failed, like this: sound/pci/hda/cs35l41_hda.c:583:12: error: ‘cs35l41_runtime_resume’ defined but not used [-Werror=unused-function] static int cs35l41_runtime_resume(struct device *dev) ^~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/cs35l41_hda.c:565:12: error: ‘cs35l41_runtime_suspend’ defined but not used [-Werror=unused-function] static int cs35l41_runtime_suspend(struct device *dev) ^~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors make[3]: *** [sound/pci/hda/cs35l41_hda.o] Error 1 commit 1a3c7bb08826 ("PM: core: Add new *_PM_OPS macros, deprecate old ones"), add new marco RUNTIME_PM_OPS to fix this unused-function problem. Fixes: 1873ebd30cc8 ("ALSA: hda: cs35l41: Support Hibernation during Suspend") Signed-off-by: Ren Zhijie <renzhijie2(a)huawei.com> --- sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 28798d5c1cf1..93cf039abb02 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1439,7 +1439,7 @@ void cs35l41_hda_remove(struct device *dev) EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); const struct dev_pm_ops cs35l41_hda_pm_ops = { - SET_RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) + RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) }; EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41); -- 2.17.1

2 1

[PATCH] usb/hiface: fix repeated words in comments
by wangjianli 25 Jul '22

25 Jul '22

Delete the redundant word 'in'. Signed-off-by: wangjianli <wangjianli(a)cdjrlc.com> --- sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 71f17f02f341..cf650fab54d7 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -225,7 +225,7 @@ static int hiface_pcm_stream_start(struct pcm_runtime *rt) } } - /* wait for first out urb to return (sent in in urb handler) */ + /* wait for first out urb to return (sent in urb handler) */ wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, HZ); if (rt->stream_wait_cond) { -- 2.36.1

2 1

[PATCH] usb/6fire: fix repeated words in comments
by wangjianli 25 Jul '22

25 Jul '22

Delete the redundant word 'in'. Signed-off-by: wangjianli <wangjianli(a)cdjrlc.com> --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 7168f1c6a37a..32c39d8bd2e5 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -175,7 +175,7 @@ static int usb6fire_pcm_stream_start(struct pcm_runtime *rt) } } - /* wait for first out urb to return (sent in in urb handler) */ + /* wait for first out urb to return (sent in urb handler) */ wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, HZ); if (rt->stream_wait_cond) -- 2.36.1

2 1

[PATCH] pci/asihpi: fix repeated words in comments
by wangjianli 25 Jul '22

25 Jul '22

Delete the redundant word 'in'. Signed-off-by: wangjianli <wangjianli(a)cdjrlc.com> --- sound/pci/asihpi/hpi6000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index aa4d06353126..88d902997b74 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -388,7 +388,7 @@ void HPI_6000(struct hpi_message *phm, struct hpi_response *phr) /* SUBSYSTEM */ /* create an adapter object and initialise it based on resource information - * passed in in the message + * passed in the message * NOTE - you cannot use this function AND the FindAdapters function at the * same time, the application must use only one of them to get the adapters */ -- 2.36.1

2 1

[PATCH] docs: driver-api: firmware: add driver firmware guidelines. (v3)
by Dave Airlie 24 Jul '22

24 Jul '22

From: Dave Airlie <airlied(a)redhat.com> A recent snafu where Intel ignored upstream feedback on a firmware change, led to a late rc6 fix being required. In order to avoid this in the future we should document some expectations around linux-firmware. I was originally going to write this for drm, but it seems quite generic advice. v2: rewritten with suggestions from Thorsten Leemhuis v3: rewritten with suggestions from Mauro Acked-by: Luis Chamberlain <mcgrof(a)kernel.org> Acked-by: Rodrigo Vivi <rodrigo.vivi(a)intel.com> Acked-by: Daniel Vetter <daniel(a)ffwll.ch> Acked-by: Harry Wentland <harry.wentland(a)amd.com> Signed-off-by: Dave Airlie <airlied(a)redhat.com> --- Documentation/driver-api/firmware/core.rst | 1 + .../firmware/firmware-usage-guidelines.rst | 44 +++++++++++++++++++ 2 files changed, 45 insertions(+) create mode 100644 Documentation/driver-api/firmware/firmware-usage-guidelines.rst diff --git a/Documentation/driver-api/firmware/core.rst b/Documentation/driver-api/firmware/core.rst index 1d1688cbc078..803cd574bbd7 100644 --- a/Documentation/driver-api/firmware/core.rst +++ b/Documentation/driver-api/firmware/core.rst @@ -13,4 +13,5 @@ documents these features. direct-fs-lookup fallback-mechanisms lookup-order + firmware-usage-guidelines diff --git a/Documentation/driver-api/firmware/firmware-usage-guidelines.rst b/Documentation/driver-api/firmware/firmware-usage-guidelines.rst new file mode 100644 index 000000000000..fdcfce42c6d2 --- /dev/null +++ b/Documentation/driver-api/firmware/firmware-usage-guidelines.rst @@ -0,0 +1,44 @@ +=================== +Firmware Guidelines +=================== + +Users switching to a newer kernel should *not* have to install newer +firmware files to keep their hardware working. At the same time updated +firmware files must not cause any regressions for users of older kernel +releases. + +Drivers that use firmware from linux-firmware should follow the rules in +this guide. (Where there is limited control of the firmware, +i.e. company doesn't support Linux, firmwares sourced from misc places, +then of course these rules will not apply strictly.) + +* Firmware files shall be designed in a way that it allows checking for + firmware ABI version changes. It is recommended that firmware files be + versioned with at least a major/minor version. It is suggested that + the firmware files in linux-firmware be named with some device + specific name, and just the major version. The firmware version should + be stored in the firmware header, or as an exception, as part of the + firmware file name, in order to let the driver detact any non-ABI + fixes/changes. The firmware files in linux-firmware should be + overwritten with the newest compatible major version. Newer major + version firmware shall remain compatible with all kernels that load + that major number. + +* If the kernel support for the hardware is normally inactive, or the + hardware isn't available for public consumption, this can + be ignored, until the first kernel release that enables that hardware. + This means no major version bumps without the kernel retaining + backwards compatibility for the older major versions. Minor version + bumps should not introduce new features that newer kernels depend on + non-optionally. + +* If a security fix needs lockstep firmware and kernel fixes in order to + be successful, then all supported major versions in the linux-firmware + repo that are required by currently supported stable/LTS kernels, + should be updated with the security fix. The kernel patches should + detect if the firmware is new enough to declare if the security issue + is fixed. All communications around security fixes should point at + both the firmware and kernel fixes. If a security fix requires + deprecating old major versions, then this should only be done as a + last option, and be stated clearly in all communications. + -- 2.36.1

2 1

[PATCH] pci/emu10k1: fix repeated words in comments
by wangjianli 24 Jul '22

24 Jul '22

Delete the redundant word 'in'. Signed-off-by: wangjianli <wangjianli(a)cdjrlc.com> --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 9d26535f3fa3..edb3f1763719 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -324,7 +324,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst return NULL; } /* fill buffer addresses but pointers are not stored so that - * snd_free_pci_page() is not called in in synth_free() + * snd_free_pci_page() is not called in synth_free() */ idx = 0; for (page = blk->first_page; page <= blk->last_page; page++, idx++) { -- 2.36.1

2 1

[PATCH] pci/asihpi: fix repeated words in comments
by wangjianli 24 Jul '22

24 Jul '22

Delete the redundant word 'in'. Signed-off-by: wangjianli <wangjianli(a)cdjrlc.com> --- sound/pci/asihpi/hpi6205.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3d6914c64c4a..27e11b5f70b9 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -445,7 +445,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) /* SUBSYSTEM */ /** Create an adapter object and initialise it based on resource information - * passed in in the message + * passed in the message * *** NOTE - you cannot use this function AND the FindAdapters function at the * same time, the application must use only one of them to get the adapters *** */ -- 2.36.1

1 0

[PATCH] slimbus: qcom-ngd-ctrl: Add check for platform_driver_register
by williamsukatube＠163.com 23 Jul '22

23 Jul '22

From: William Dean <williamsukatube(a)gmail.com> As platform_driver_register() could fail, it should be better to deal with the return value in order to maintain the code consisitency. Fixes: 1830dad34c070 ("slimbus: ngd: register ngd driver only once.") Reported-by: Hacash Robot <hacashRobot(a)santino.com> Signed-off-by: William Dean <williamsukatube(a)gmail.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 0aa8408464ad..a31ec00a37ac 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1585,7 +1585,12 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) return PTR_ERR(pds); } - platform_driver_register(&qcom_slim_ngd_driver); + ret = platform_driver_register(&qcom_slim_ngd_driver); + if (ret) { + dev_err(dev, "platform driver register failed: %d\n", ret); + return ret; + } + return of_qcom_slim_ngd_register(dev, ctrl); } -- 2.25.1

1 0

[PATCH v2] ASoC: dt-bindings: atmel-i2s: Convert to json-schema
by Ryan.Wanner＠microchip.com 23 Jul '22

23 Jul '22

From: Ryan Wanner <Ryan.Wanner(a)microchip.com> Convert atmel i2s devicetree binding to json-schema. Change file name to match json-schema naming. Signed-off-by: Ryan Wanner <Ryan.Wanner(a)microchip.com> --- Note: running dtbs_check will fail unless updated with this patch, https://lore.kernel.org/linux-arm-kernel/20220707215812.193008-1-Ryan.Wanne… v1 -> v2: - Fix formatting for clock description. - Fix formatting for dma description. .../bindings/sound/atmel,sama5d2-i2s.yaml | 85 +++++++++++++++++++ .../devicetree/bindings/sound/atmel-i2s.txt | 46 ---------- 2 files changed, 85 insertions(+), 46 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml new file mode 100644 index 000000000000..0cd1ff89baed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml @@ -0,0 +1,85 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel I2S controller + +maintainers: + - Nicolas Ferre <nicolas.ferre(a)microchip.com> + - Alexandre Belloni <alexandre.belloni(a)bootlin.com> + - Claudiu Beznea <claudiu.beznea(a)microchip.com> + +description: + Atmel I2S (Inter-IC Sound Controller) bus is the standard + interface for connecting audio devices, such as audio codecs. + +properties: + compatible: + const: atmel,sama5d2-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + - description: Generated clock (Optional) + - description: I2S mux clock (Optional). Set + with gclk when Master Mode is required. + minItems: 1 + + clock-names: + items: + - const: pclk + - const: gclk + - const: muxclk + minItems: 1 + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f8050000 { + compatible = "atmel,sama5d2-i2s"; + reg = <0xf8050000 0x300>; + interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(31))>, + <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(32))>; + dma-names = "tx", "rx"; + clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; + clock-names = "pclk", "gclk", "muxclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s0_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-i2s.txt b/Documentation/devicetree/bindings/sound/atmel-i2s.txt deleted file mode 100644 index 40549f496a81..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-i2s.txt +++ /dev/null @@ -1,46 +0,0 @@ -* Atmel I2S controller - -Required properties: -- compatible: Should be "atmel,sama5d2-i2s". -- reg: Should be the physical base address of the controller and the - length of memory mapped region. -- interrupts: Should contain the interrupt for the controller. -- dmas: Should be one per channel name listed in the dma-names property, - as described in atmel-dma.txt and dma.txt files. -- dma-names: Two dmas have to be defined, "tx" and "rx". - This IP also supports one shared channel for both rx and tx; - if this mode is used, one "rx-tx" name must be used. -- clocks: Must contain an entry for each entry in clock-names. - Please refer to clock-bindings.txt. -- clock-names: Should be one of each entry matching the clocks phandles list: - - "pclk" (peripheral clock) Required. - - "gclk" (generated clock) Optional (1). - - "muxclk" (I2S mux clock) Optional (1). - -Optional properties: -- pinctrl-0: Should specify pin control groups used for this controller. -- princtrl-names: Should contain only one value - "default". - - -(1) : Only the peripheral clock is required. The generated clock and the I2S - mux clock are optional and should only be set together, when Master Mode - is required. - -Example: - - i2s@f8050000 { - compatible = "atmel,sama5d2-i2s"; - reg = <0xf8050000 0x300>; - interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; - dmas = <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(31))>, - <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(32))>; - dma-names = "tx", "rx"; - clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; - clock-names = "pclk", "gclk", "muxclk"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s0_default>; - }; -- 2.34.1

4 3

[PATCH] ASoC: amd: yc: Update DMI table entries
by syed sabakareem 23 Jul '22

23 Jul '22

Removed intel DMI product id's 21AW/21AX/21D8/21D9/21BN/21BQ in DMI table and updated DMI entry for AMD platform X13 Gen 3 platform 21CM/21CN. Link: https://bugzilla.kernel.org/show_bug.cgi?id=216267 Signed-off-by: syed sabakareem <Syed.SabaKareem(a)amd.com> Reviewed-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 32 ++------------------------------ 1 file changed, 2 insertions(+), 30 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index f06e6c1a7799..ecfe7a790790 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -105,28 +105,14 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "21AW"), + DMI_MATCH(DMI_PRODUCT_NAME, "21CM"), } }, { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "21AX"), - } - }, - { - .driver_data = &acp6x_card, - .matches = { - DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "21BN"), - } - }, - { - .driver_data = &acp6x_card, - .matches = { - DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "21BQ"), + DMI_MATCH(DMI_PRODUCT_NAME, "21CN"), } }, { @@ -157,20 +143,6 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, - { - .driver_data = &acp6x_card, - .matches = { - DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "21D8"), - } - }, - { - .driver_data = &acp6x_card, - .matches = { - DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "21D9"), - } - }, {} }; -- 2.25.1

4 4

[PATCH] ASoC: audio-graph-card2: Add of_node_put() in fail path
by Liang He 23 Jul '22

23 Jul '22

In asoc_simple_parse_dai(), we should call of_node_put() for the reference returned by of_graph_get_port_parent() in fail path. Fixes: 6e5f68fe3f2d ("ASoC: add Audio Graph Card2 driver") Signed-off-by: Liang He <windhl(a)126.com> --- sound/soc/generic/audio-graph-card2.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 18d053a0d05c..8ac6df645ee6 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -445,8 +445,10 @@ static int asoc_simple_parse_dai(struct device_node *ep, * if he unbinded CPU or Codec. */ ret = snd_soc_get_dai_name(&args, &dlc->dai_name); - if (ret < 0) + if (ret < 0) { + of_node_put(node); return ret; + } dlc->of_node = node; -- 2.25.1

2 1

[PATCH v6 2/4] PCI: Move isa_dma_bridge_buggy out of dma.h
by Stafford Horne 22 Jul '22

22 Jul '22

During recent PCI cleanups we noticed that the isa_dma_bridge_buggy symbol supported by all architectures is actually only used for x86_32. This patch moves the symbol out of all architectures limiting usage to only x86_32. This is possible because only x86_32 platforms or quirks existing in PCI devices supported on x86_32 ever set this. A new global header linux/isa-dma.h is added to provide a common place to maintain the definition. Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Suggested-by: Christoph Hellwig <hch(a)infradead.org> Acked-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reviewed-by: Christoph Hellwig <hch(a)lst.de> Signed-off-by: Stafford Horne <shorne(a)gmail.com> --- arch/alpha/include/asm/dma.h | 9 --------- arch/arc/include/asm/dma.h | 5 ----- arch/arm/include/asm/dma.h | 6 ------ arch/arm64/include/asm/pci.h | 2 -- arch/csky/include/asm/pci.h | 2 -- arch/ia64/include/asm/dma.h | 2 -- arch/m68k/include/asm/dma.h | 6 ------ arch/microblaze/include/asm/dma.h | 6 ------ arch/mips/include/asm/dma.h | 8 -------- arch/parisc/include/asm/dma.h | 6 ------ arch/powerpc/include/asm/dma.h | 6 ------ arch/riscv/include/asm/pci.h | 2 -- arch/s390/include/asm/dma.h | 6 ------ arch/sh/include/asm/dma.h | 6 ------ arch/sparc/include/asm/dma.h | 8 -------- arch/um/include/asm/pci.h | 2 -- arch/x86/include/asm/dma.h | 8 -------- arch/xtensa/include/asm/dma.h | 7 ------- drivers/comedi/drivers/comedi_isadma.c | 2 +- drivers/pci/pci.c | 2 ++ drivers/pci/quirks.c | 4 +++- include/linux/isa-dma.h | 14 ++++++++++++++ sound/core/isadma.c | 2 +- 23 files changed, 21 insertions(+), 100 deletions(-) create mode 100644 include/linux/isa-dma.h diff --git a/arch/alpha/include/asm/dma.h b/arch/alpha/include/asm/dma.h index 28610ea7786d..a04d76b96089 100644 --- a/arch/alpha/include/asm/dma.h +++ b/arch/alpha/include/asm/dma.h @@ -365,13 +365,4 @@ extern void free_dma(unsigned int dmanr); /* release it again */ #define KERNEL_HAVE_CHECK_DMA extern int check_dma(unsigned int dmanr); -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - - #endif /* _ASM_DMA_H */ diff --git a/arch/arc/include/asm/dma.h b/arch/arc/include/asm/dma.h index 5b744f4b10a7..02431027ed2f 100644 --- a/arch/arc/include/asm/dma.h +++ b/arch/arc/include/asm/dma.h @@ -7,10 +7,5 @@ #define ASM_ARC_DMA_H #define MAX_DMA_ADDRESS 0xC0000000 -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy 0 -#endif #endif diff --git a/arch/arm/include/asm/dma.h b/arch/arm/include/asm/dma.h index a81dda65c576..907d139be431 100644 --- a/arch/arm/include/asm/dma.h +++ b/arch/arm/include/asm/dma.h @@ -143,10 +143,4 @@ extern int get_dma_residue(unsigned int chan); #endif /* CONFIG_ISA_DMA_API */ -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __ASM_ARM_DMA_H */ diff --git a/arch/arm64/include/asm/pci.h b/arch/arm64/include/asm/pci.h index 0aebc3488c32..682c922b5658 100644 --- a/arch/arm64/include/asm/pci.h +++ b/arch/arm64/include/asm/pci.h @@ -20,8 +20,6 @@ #define arch_can_pci_mmap_wc() 1 #define ARCH_GENERIC_PCI_MMAP_RESOURCE 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/csky/include/asm/pci.h b/arch/csky/include/asm/pci.h index 0535f1aaae38..5c02454ec724 100644 --- a/arch/csky/include/asm/pci.h +++ b/arch/csky/include/asm/pci.h @@ -15,8 +15,6 @@ /* C-SKY shim does not initialize PCI bus */ #define pcibios_assign_all_busses() 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/ia64/include/asm/dma.h b/arch/ia64/include/asm/dma.h index 59625e9c1f9c..eaed2626ffda 100644 --- a/arch/ia64/include/asm/dma.h +++ b/arch/ia64/include/asm/dma.h @@ -12,8 +12,6 @@ extern unsigned long MAX_DMA_ADDRESS; -extern int isa_dma_bridge_buggy; - #define free_dma(x) #endif /* _ASM_IA64_DMA_H */ diff --git a/arch/m68k/include/asm/dma.h b/arch/m68k/include/asm/dma.h index f6c5e0dfb4e5..1c8d9c5bc2fa 100644 --- a/arch/m68k/include/asm/dma.h +++ b/arch/m68k/include/asm/dma.h @@ -6,10 +6,4 @@ bootmem allocator (but this should do it for this) */ #define MAX_DMA_ADDRESS PAGE_OFFSET -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _M68K_DMA_H */ diff --git a/arch/microblaze/include/asm/dma.h b/arch/microblaze/include/asm/dma.h index f801582be912..7484c9eb66c4 100644 --- a/arch/microblaze/include/asm/dma.h +++ b/arch/microblaze/include/asm/dma.h @@ -9,10 +9,4 @@ /* Virtual address corresponding to last available physical memory address. */ #define MAX_DMA_ADDRESS (CONFIG_KERNEL_START + memory_size - 1) -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_MICROBLAZE_DMA_H */ diff --git a/arch/mips/include/asm/dma.h b/arch/mips/include/asm/dma.h index be726b943530..d6186e6bea7e 100644 --- a/arch/mips/include/asm/dma.h +++ b/arch/mips/include/asm/dma.h @@ -307,12 +307,4 @@ static __inline__ int get_dma_residue(unsigned int dmanr) extern int request_dma(unsigned int dmanr, const char * device_id); /* reserve a DMA channel */ extern void free_dma(unsigned int dmanr); /* release it again */ -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_DMA_H */ diff --git a/arch/parisc/include/asm/dma.h b/arch/parisc/include/asm/dma.h index eea80ed34e6d..9e8c101de902 100644 --- a/arch/parisc/include/asm/dma.h +++ b/arch/parisc/include/asm/dma.h @@ -176,10 +176,4 @@ static __inline__ void set_dma_count(unsigned int dmanr, unsigned int count) #define free_dma(dmanr) -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_DMA_H */ diff --git a/arch/powerpc/include/asm/dma.h b/arch/powerpc/include/asm/dma.h index 6161a9596196..d97c66d9ae34 100644 --- a/arch/powerpc/include/asm/dma.h +++ b/arch/powerpc/include/asm/dma.h @@ -340,11 +340,5 @@ extern int request_dma(unsigned int dmanr, const char *device_id); /* release it again */ extern void free_dma(unsigned int dmanr); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __KERNEL__ */ #endif /* _ASM_POWERPC_DMA_H */ diff --git a/arch/riscv/include/asm/pci.h b/arch/riscv/include/asm/pci.h index a7b8f0d0df7f..f904df586c03 100644 --- a/arch/riscv/include/asm/pci.h +++ b/arch/riscv/include/asm/pci.h @@ -20,8 +20,6 @@ #define ARCH_GENERIC_PCI_MMAP_RESOURCE 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/s390/include/asm/dma.h b/arch/s390/include/asm/dma.h index 6f26f35d4a71..dec1c4ce628c 100644 --- a/arch/s390/include/asm/dma.h +++ b/arch/s390/include/asm/dma.h @@ -11,10 +11,4 @@ */ #define MAX_DMA_ADDRESS 0x80000000 -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_S390_DMA_H */ diff --git a/arch/sh/include/asm/dma.h b/arch/sh/include/asm/dma.h index 17d23ae98c77..c8bee3f985a2 100644 --- a/arch/sh/include/asm/dma.h +++ b/arch/sh/include/asm/dma.h @@ -137,10 +137,4 @@ extern int register_chan_caps(const char *dmac, struct dma_chan_caps *capslist); extern int dma_create_sysfs_files(struct dma_channel *, struct dma_info *); extern void dma_remove_sysfs_files(struct dma_channel *, struct dma_info *); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __ASM_SH_DMA_H */ diff --git a/arch/sparc/include/asm/dma.h b/arch/sparc/include/asm/dma.h index 462e7c794a09..08043f35b110 100644 --- a/arch/sparc/include/asm/dma.h +++ b/arch/sparc/include/asm/dma.h @@ -82,14 +82,6 @@ #define DMA_BURST64 0x40 #define DMA_BURSTBITS 0x7f -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #ifdef CONFIG_SPARC32 struct device; diff --git a/arch/um/include/asm/pci.h b/arch/um/include/asm/pci.h index 26b96c02ef61..1211855aff34 100644 --- a/arch/um/include/asm/pci.h +++ b/arch/um/include/asm/pci.h @@ -9,8 +9,6 @@ #define pcibios_assign_all_busses() 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI_DOMAINS static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/x86/include/asm/dma.h b/arch/x86/include/asm/dma.h index 8e95aa4b0d17..8ae6e0e11b8b 100644 --- a/arch/x86/include/asm/dma.h +++ b/arch/x86/include/asm/dma.h @@ -307,12 +307,4 @@ extern int request_dma(unsigned int dmanr, const char *device_id); extern void free_dma(unsigned int dmanr); #endif -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_X86_DMA_H */ diff --git a/arch/xtensa/include/asm/dma.h b/arch/xtensa/include/asm/dma.h index bb099a373b5a..172644539032 100644 --- a/arch/xtensa/include/asm/dma.h +++ b/arch/xtensa/include/asm/dma.h @@ -52,11 +52,4 @@ extern int request_dma(unsigned int dmanr, const char * device_id); extern void free_dma(unsigned int dmanr); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - - #endif diff --git a/drivers/comedi/drivers/comedi_isadma.c b/drivers/comedi/drivers/comedi_isadma.c index 700982464c53..020b3d1e1ac0 100644 --- a/drivers/comedi/drivers/comedi_isadma.c +++ b/drivers/comedi/drivers/comedi_isadma.c @@ -8,7 +8,7 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/dma-mapping.h> -#include <asm/dma.h> +#include <linux/isa-dma.h> #include <linux/comedi/comedidev.h> #include <linux/comedi/comedi_isadma.h> diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index cfaf40a540a8..60c55d2cb2cc 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -41,8 +41,10 @@ const char *pci_power_names[] = { }; EXPORT_SYMBOL_GPL(pci_power_names); +#ifdef CONFIG_X86_32 int isa_dma_bridge_buggy; EXPORT_SYMBOL(isa_dma_bridge_buggy); +#endif int pci_pci_problems; EXPORT_SYMBOL(pci_pci_problems); diff --git a/drivers/pci/quirks.c b/drivers/pci/quirks.c index 41aeaa235132..6fc64509eee7 100644 --- a/drivers/pci/quirks.c +++ b/drivers/pci/quirks.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/export.h> #include <linux/pci.h> +#include <linux/isa-dma.h> /* isa_dma_bridge_buggy */ #include <linux/init.h> #include <linux/delay.h> #include <linux/acpi.h> @@ -30,7 +31,6 @@ #include <linux/pm_runtime.h> #include <linux/suspend.h> #include <linux/switchtec.h> -#include <asm/dma.h> /* isa_dma_bridge_buggy */ #include "pci.h" static ktime_t fixup_debug_start(struct pci_dev *dev, @@ -239,6 +239,7 @@ static void quirk_passive_release(struct pci_dev *dev) DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82441, quirk_passive_release); DECLARE_PCI_FIXUP_RESUME(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82441, quirk_passive_release); +#ifdef CONFIG_X86_32 /* * The VIA VP2/VP3/MVP3 seem to have some 'features'. There may be a * workaround but VIA don't answer queries. If you happen to have good @@ -265,6 +266,7 @@ DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M1533, quirk_isa_dma DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_1, quirk_isa_dma_hangs); DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_2, quirk_isa_dma_hangs); DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_3, quirk_isa_dma_hangs); +#endif /* * Intel NM10 "TigerPoint" LPC PM1a_STS.BM_STS must be clear diff --git a/include/linux/isa-dma.h b/include/linux/isa-dma.h new file mode 100644 index 000000000000..61504a8c1b9e --- /dev/null +++ b/include/linux/isa-dma.h @@ -0,0 +1,14 @@ +/* SPDX-License-Identifier: GPL-2.0 */ + +#ifndef __LINUX_ISA_DMA_H +#define __LINUX_ISA_DMA_H + +#include <asm/dma.h> + +#if defined(CONFIG_PCI) && defined(CONFIG_X86_32) +extern int isa_dma_bridge_buggy; +#else +#define isa_dma_bridge_buggy (0) +#endif + +#endif /* __LINUX_ISA_DMA_H */ diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 1f45ede023b4..18a86212e3a8 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -12,8 +12,8 @@ #undef HAVE_REALLY_SLOW_DMA_CONTROLLER #include <linux/export.h> +#include <linux/isa-dma.h> #include <sound/core.h> -#include <asm/dma.h> /** * snd_dma_program - program an ISA DMA transfer -- 2.36.1

1 0

[PATCH 1/2] firmware: cs_dsp: Add pre_stop callback
by Charles Keepax 22 Jul '22

22 Jul '22

The code already has a post_stop callback, add a matching pre_stop callback to the client_ops that is called before execution is stopped. This callback provides a convenient place for the client code to communicate with the DSP before it is stopped. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 3 +++ include/linux/firmware/cirrus/cs_dsp.h | 4 +++- 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 7dad6f57d9704..b402f841d72cf 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -2725,6 +2725,9 @@ void cs_dsp_stop(struct cs_dsp *dsp) mutex_lock(&dsp->pwr_lock); + if (dsp->client_ops->pre_stop) + dsp->client_ops->pre_stop(dsp); + dsp->running = false; if (dsp->ops->stop_core) diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 30055706cce21..6ab230218df0e 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -189,7 +189,8 @@ struct cs_dsp { * @control_remove: Called under the pwr_lock when a control is destroyed * @pre_run: Called under the pwr_lock by cs_dsp_run() before the core is started * @post_run: Called under the pwr_lock by cs_dsp_run() after the core is started - * @post_stop: Called under the pwr_lock by cs_dsp_stop() + * @pre_stop: Called under the pwr_lock by cs_dsp_stop() before the core is stopped + * @post_stop: Called under the pwr_lock by cs_dsp_stop() after the core is stopped * @watchdog_expired: Called when a watchdog expiry is detected * * These callbacks give the cs_dsp client an opportunity to respond to events @@ -200,6 +201,7 @@ struct cs_dsp_client_ops { void (*control_remove)(struct cs_dsp_coeff_ctl *ctl); int (*pre_run)(struct cs_dsp *dsp); int (*post_run)(struct cs_dsp *dsp); + void (*pre_stop)(struct cs_dsp *dsp); void (*post_stop)(struct cs_dsp *dsp); void (*watchdog_expired)(struct cs_dsp *dsp); }; -- 2.30.2

2 2

[PATCH v2] ASoC: amd: vangogh: Use non-legacy DAI naming for cs35l41
by Cristian Ciocaltea 22 Jul '22

22 Jul '22

Unlike most CODEC drivers, the CS35L41 driver did not have the non_legacy_dai_naming set, meaning the corresponding DAI has been traditionally registered using the legacy naming: spi-VLV1776:0x The recent migration to the new legacy DAI naming style has implicitly corrected that behavior and DAI gets now registered via the non-legacy naming, i.e. cs35l41-pcm. The problem is the acp5x platform driver is now broken as it continues to refer to the above mentioned codec using the legacy DAI naming in function acp5x_cs35l41_hw_params() and, therefore, the related setup is not being executed anymore. Let's fix that by replacing the obsolete DAI name with the correct one. Fixes: 129f055a2144 ("ASoC: core: Switch core to new DAI naming flag") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes in v2: - Corrected the SHA for the Fixes commit - Added Reviewed-by tag from Charles sound/soc/amd/vangogh/acp5x-mach.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index 727de46860b1..af3737ef9707 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -178,8 +178,7 @@ static int acp5x_cs35l41_hw_params(struct snd_pcm_substream *substream, ret = 0; for (i = 0; i < num_codecs; i++) { codec_dai = asoc_rtd_to_codec(rtd, i); - if ((strcmp(codec_dai->name, "spi-VLV1776:00") == 0) || - (strcmp(codec_dai->name, "spi-VLV1776:01") == 0)) { + if (strcmp(codec_dai->name, "cs35l41-pcm") == 0) { switch (params_rate(params)) { case 48000: bclk_val = 1536000; -- 2.37.1

2 1

[PATCH] ASoC: Intel: avs: Use lookup table to create modules
by Amadeusz Sławiński 22 Jul '22

22 Jul '22

As reported by Nathan, when building avs driver using clang with: CONFIG_COMPILE_TEST=y CONFIG_FORTIFY_SOURCE=y CONFIG_KASAN=y CONFIG_PCI=y CONFIG_SOUND=y CONFIG_SND=y CONFIG_SND_SOC=y CONFIG_SND_SOC_INTEL_AVS=y there are reports of too big stack use, like: sound/soc/intel/avs/path.c:815:18: error: stack frame size (2176) exceeds limit (2048) in 'avs_path_create' [-Werror,-Wframe-larger-than] struct avs_path *avs_path_create(struct avs_dev *adev, u32 dma_id, ^ 1 error generated. This is apparently caused by inlining many calls to guid_equal which inlines fortified memcpy, using 2 size_t variables. Instead of hardcoding many calls to guid_equal, use lookup table with one call, this improves stack usage. Link: https://lore.kernel.org/alsa-devel/YtlzY9aYdbS4Y3+l@dev-arch.thelio-3990X/T/ Link: https://github.com/ClangBuiltLinux/linux/issues/1642 Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reported-by: Nathan Chancellor <nathan(a)kernel.org> Build-tested-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/intel/avs/path.c | 54 ++++++++++++++++++++------------------ 1 file changed, 29 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 3d46dd5e5bc4..ce157a8d6552 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -449,35 +449,39 @@ static int avs_modext_create(struct avs_dev *adev, struct avs_path_module *mod) return ret; } +static int avs_probe_create(struct avs_dev *adev, struct avs_path_module *mod) +{ + dev_err(adev->dev, "Probe module can't be instantiated by topology"); + return -EINVAL; +} + +struct avs_module_create { + guid_t *guid; + int (*create)(struct avs_dev *adev, struct avs_path_module *mod); +}; + +static struct avs_module_create avs_module_create[] = { + { &AVS_MIXIN_MOD_UUID, avs_modbase_create }, + { &AVS_MIXOUT_MOD_UUID, avs_modbase_create }, + { &AVS_KPBUFF_MOD_UUID, avs_modbase_create }, + { &AVS_COPIER_MOD_UUID, avs_copier_create }, + { &AVS_MICSEL_MOD_UUID, avs_micsel_create }, + { &AVS_MUX_MOD_UUID, avs_mux_create }, + { &AVS_UPDWMIX_MOD_UUID, avs_updown_mix_create }, + { &AVS_SRCINTC_MOD_UUID, avs_src_create }, + { &AVS_AEC_MOD_UUID, avs_aec_create }, + { &AVS_ASRC_MOD_UUID, avs_asrc_create }, + { &AVS_INTELWOV_MOD_UUID, avs_wov_create }, + { &AVS_PROBE_MOD_UUID, avs_probe_create }, +}; + static int avs_path_module_type_create(struct avs_dev *adev, struct avs_path_module *mod) { const guid_t *type = &mod->template->cfg_ext->type; - if (guid_equal(type, &AVS_MIXIN_MOD_UUID) || - guid_equal(type, &AVS_MIXOUT_MOD_UUID) || - guid_equal(type, &AVS_KPBUFF_MOD_UUID)) - return avs_modbase_create(adev, mod); - if (guid_equal(type, &AVS_COPIER_MOD_UUID)) - return avs_copier_create(adev, mod); - if (guid_equal(type, &AVS_MICSEL_MOD_UUID)) - return avs_micsel_create(adev, mod); - if (guid_equal(type, &AVS_MUX_MOD_UUID)) - return avs_mux_create(adev, mod); - if (guid_equal(type, &AVS_UPDWMIX_MOD_UUID)) - return avs_updown_mix_create(adev, mod); - if (guid_equal(type, &AVS_SRCINTC_MOD_UUID)) - return avs_src_create(adev, mod); - if (guid_equal(type, &AVS_AEC_MOD_UUID)) - return avs_aec_create(adev, mod); - if (guid_equal(type, &AVS_ASRC_MOD_UUID)) - return avs_asrc_create(adev, mod); - if (guid_equal(type, &AVS_INTELWOV_MOD_UUID)) - return avs_wov_create(adev, mod); - - if (guid_equal(type, &AVS_PROBE_MOD_UUID)) { - dev_err(adev->dev, "Probe module can't be instantiated by topology"); - return -EINVAL; - } + for (int i = 0; i < ARRAY_SIZE(avs_module_create); i++) + if (guid_equal(type, avs_module_create[i].guid)) + return avs_module_create[i].create(adev, mod); return avs_modext_create(adev, mod); } -- 2.25.1

3 2

[PATCH] ALSA: usb-audio: Add quirk for Behringer UMC202HD
by Takashi Iwai 22 Jul '22

22 Jul '22

Just like other Behringer models, UMC202HD (USB ID 1397:0507) requires the quirk for the stable streaming, too. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 968d90caeefa..168fd802d70b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1843,6 +1843,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1397, 0x0507, /* Behringer UMC202HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x1397, 0x0508, /* Behringer UMC204HD */ QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x1397, 0x0509, /* Behringer UMC404HD */ -- 2.35.3

1 0

[PATCH 1/5] ASoC: amd: acp: Modify local variables name to generic
by Venkata Prasad Potturu 22 Jul '22

22 Jul '22

Change local variables name to be generic in irq handler. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-platform.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 327e17736dbd..b1ca52274375 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -94,19 +94,19 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; u16 i2s_flag = 0; - u32 val, val1, i; + u32 ext_intr_stat, ext_intr_stat1, i; if (!adata) return IRQ_NONE; if (adata->rsrc->no_of_ctrls == 2) - val1 = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); + ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); - val = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); + ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); for (i = 0; i < ACP_MAX_STREAM; i++) { stream = adata->stream[i]; - if (stream && (val & stream->irq_bit)) { + if (stream && (ext_intr_stat & stream->irq_bit)) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); @@ -114,7 +114,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) break; } if (adata->rsrc->no_of_ctrls == 2) { - if (stream && (val1 & stream->irq_bit)) { + if (stream && (ext_intr_stat1 & stream->irq_bit)) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); -- 2.25.1

3 7

[PATCH 1/5] ASoC: amd: add RPL Platform acp header file
by syed sabakareem 22 Jul '22

22 Jul '22

From: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> Add ACP register header file for RPL platform. Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- .../soc/amd/rpl/rpl_acp6x_chip_offset_byte.h | 30 +++++++++++++++++++ 1 file changed, 30 insertions(+) create mode 100644 sound/soc/amd/rpl/rpl_acp6x_chip_offset_byte.h diff --git a/sound/soc/amd/rpl/rpl_acp6x_chip_offset_byte.h b/sound/soc/amd/rpl/rpl_acp6x_chip_offset_byte.h new file mode 100644 index 000000000000..456498f5396d --- /dev/null +++ b/sound/soc/amd/rpl/rpl_acp6x_chip_offset_byte.h @@ -0,0 +1,30 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * AMD ACP 6.2 Register Documentation + * + * Copyright 2022 Advanced Micro Devices, Inc. + */ + +#ifndef _rpl_acp6x_OFFSET_HEADER +#define _rpl_acp6x_OFFSET_HEADER + +/* Registers from ACP_CLKRST block */ +#define ACP_SOFT_RESET 0x1241000 +#define ACP_CONTROL 0x1241004 +#define ACP_STATUS 0x1241008 +#define ACP_DYNAMIC_CG_MASTER_CONTROL 0x1241010 +#define ACP_PGFSM_CONTROL 0x124101C +#define ACP_PGFSM_STATUS 0x1241020 +#define ACP_CLKMUX_SEL 0x1241024 + +/* Registers from ACP_AON block */ +#define ACP_PME_EN 0x1241400 +#define ACP_DEVICE_STATE 0x1241404 +#define AZ_DEVICE_STATE 0x1241408 +#define ACP_PIN_CONFIG 0x1241440 +#define ACP_PAD_PULLUP_CTRL 0x1241444 +#define ACP_PAD_PULLDOWN_CTRL 0x1241448 +#define ACP_PAD_DRIVE_STRENGTH_CTRL 0x124144C +#define ACP_PAD_SCHMEN_CTRL 0x1241450 + +#endif -- 2.25.1

4 9

[PATCH] ASoC: audio-graph-card: Add of_node_put() in fail path
by Liang He 22 Jul '22

22 Jul '22

In asoc_simple_parse_dai(), we should call of_node_put() for the reference returned by of_graph_get_port_parent() in fail path. Fixes: ae30a694da4c ("ASoC: simple-card-utils: add asoc_simple_card_parse_dai()") Signed-off-by: Liang He <windhl(a)126.com> --- sound/soc/generic/audio-graph-card.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2b598af8feef..b327372f2e4a 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -158,8 +158,10 @@ static int asoc_simple_parse_dai(struct device_node *ep, * if he unbinded CPU or Codec. */ ret = snd_soc_get_dai_name(&args, &dlc->dai_name); - if (ret < 0) + if (ret < 0) { + of_node_put(node); return ret; + } dlc->of_node = node; -- 2.25.1

2 1

[PATCH v2 -next 0/5] ASoC: fsl: Fix sparse warning
by Shengjiu Wang 22 Jul '22

22 Jul '22

Fix sparse warning changes in v2: - use pcm_format_to_bits - use u32 asrc_fmt, then convert it to snd_pcm_format_t Shengjiu Wang (5): ASoC: fsl_sai: Don't use plain integer as NULL pointer ASoC: fsl_asrc: force cast the asrc_format type ASoC: fsl-asoc-card: force cast the asrc_format type ASoC: fsl_easrc: use snd_pcm_format_t type for sample_format ASoC: imx-card: use snd_pcm_format_t type for asrc_format sound/soc/fsl/fsl-asoc-card.c | 5 +++-- sound/soc/fsl/fsl_asrc.c | 6 ++++-- sound/soc/fsl/fsl_easrc.c | 9 ++++++--- sound/soc/fsl/fsl_easrc.h | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/imx-card.c | 8 +++++--- 6 files changed, 20 insertions(+), 12 deletions(-) -- 2.34.1

2 6

[PATCH] ASoC: qcom: q6dsp: Fix an off-by-one in q6adm_alloc_copp()
by Christophe JAILLET 22 Jul '22

22 Jul '22

find_first_zero_bit() returns MAX_COPPS_PER_PORT at max here. So 'idx' should be tested with ">=" or the test can't match. Fixes: 7b20b2be51e1 ("ASoC: qdsp6: q6adm: Add q6adm driver") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/qcom/qdsp6/q6adm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 01f383888b62..1530e98df165 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -217,7 +217,7 @@ static struct q6copp *q6adm_alloc_copp(struct q6adm *adm, int port_idx) idx = find_first_zero_bit(&adm->copp_bitmap[port_idx], MAX_COPPS_PER_PORT); - if (idx > MAX_COPPS_PER_PORT) + if (idx >= MAX_COPPS_PER_PORT) return ERR_PTR(-EBUSY); c = kzalloc(sizeof(*c), GFP_ATOMIC); -- 2.34.1

3 6

[PATCH] ASoC: amd: vangogh: Use non-legacy DAI naming for cs35l41
by Cristian Ciocaltea 22 Jul '22

22 Jul '22

Unlike most CODEC drivers, the CS35L41 driver did not have the non_legacy_dai_naming set, meaning the corresponding DAI has been traditionally registered using the legacy naming: spi-VLV1776:0x The recent migration to the new legacy DAI naming style has implicitly corrected that behavior and DAI gets now registered via the non-legacy naming, i.e. cs35l41-pcm. The problem is the acp5x platform driver is now broken as it continues to refer to the above mentioned codec using the legacy DAI naming in function acp5x_cs35l41_hw_params() and, therefore, the related setup is not being executed anymore. Let's fix that by replacing the obsolete DAI name with the correct one. Fixes: bc949a3b4af3 ("ASoC: core: Switch core to new DAI naming flag") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- sound/soc/amd/vangogh/acp5x-mach.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index 727de46860b1..af3737ef9707 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -178,8 +178,7 @@ static int acp5x_cs35l41_hw_params(struct snd_pcm_substream *substream, ret = 0; for (i = 0; i < num_codecs; i++) { codec_dai = asoc_rtd_to_codec(rtd, i); - if ((strcmp(codec_dai->name, "spi-VLV1776:00") == 0) || - (strcmp(codec_dai->name, "spi-VLV1776:01") == 0)) { + if (strcmp(codec_dai->name, "cs35l41-pcm") == 0) { switch (params_rate(params)) { case 48000: bclk_val = 1536000; -- 2.37.1

2 3

[PATCH] ALSA: asihpi: Fix typo in comments
by shaomin Deng 22 Jul '22

22 Jul '22

Delete the repeated word "in" in comments. Signed-off-by: shaomin Deng <dengshaomin(a)cdjrlc.com> --- sound/pci/asihpi/hpi6205.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3d6914c64c4a..27e11b5f70b9 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -445,7 +445,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) /* SUBSYSTEM */ /** Create an adapter object and initialise it based on resource information - * passed in in the message + * passed in the message * *** NOTE - you cannot use this function AND the FindAdapters function at the * same time, the application must use only one of them to get the adapters *** */ -- 2.35.1

2 1

[PATCH] ALSA: emu10k1: Fix typo in comments
by shaomin Deng 22 Jul '22

22 Jul '22

Remove the rebundant word "in" in comments. Signed-off-by: shaomin Deng <dengshaomin(a)cdjrlc.com> --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 9d26535f3fa3..edb3f1763719 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -324,7 +324,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst return NULL; } /* fill buffer addresses but pointers are not stored so that - * snd_free_pci_page() is not called in in synth_free() + * snd_free_pci_page() is not called in synth_free() */ idx = 0; for (page = blk->first_page; page <= blk->last_page; page++, idx++) { -- 2.35.1

2 1

[PATCH] ASoC: cs35l41: Set the new legacy DAI naming flag
by Cristian Ciocaltea 22 Jul '22

22 Jul '22

Unlike most CODEC drivers, the CS35L41 driver did not have the non_legacy_dai_naming set, meaning it uses the legacy naming. The recent migration to the new legacy DAI naming style has broken driver functionality because it is now expected to set the new legacy DAI naming flag in order to instruct the core subsystem to use the legacy name format on DAI registration. Let's fix this by setting the legacy_dai_naming flag accordingly. Fixes: bc949a3b4af3 ("ASoC: core: Switch core to new DAI naming flag") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- sound/soc/codecs/cs35l41.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index c223d83e02cf..b49f9a32abf2 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1028,6 +1028,7 @@ static const struct snd_soc_component_driver soc_component_dev_cs35l41 = { .set_sysclk = cs35l41_component_set_sysclk, .endianness = 1, + .legacy_dai_naming = 1, }; static int cs35l41_handle_pdata(struct device *dev, struct cs35l41_hw_cfg *hw_cfg) -- 2.37.1

2 3

[PATCH] ASoC: Intel: avs: Mark avs_path_module_type_create() as noinline
by Nathan Chancellor 21 Jul '22

21 Jul '22

When building ARCH=arm64 allmodconfig with clang, there is a warning about high stack usage in avs_path_create(), which breaks the build due to CONFIG_WERROR=y: sound/soc/intel/avs/path.c:815:18: error: stack frame size (2176) exceeds limit (2048) in 'avs_path_create' [-Werror,-Wframe-larger-than] struct avs_path *avs_path_create(struct avs_dev *adev, u32 dma_id, ^ 1 error generated. This warning is also visible with allmodconfig on other architectures. The minimum set of configs that triggers this on top of ARCH=arm64 allnoconfig: CONFIG_COMPILE_TEST=y CONFIG_FORTIFY_SOURCE=y CONFIG_KASAN=y CONFIG_PCI=y CONFIG_SOUND=y CONFIG_SND=y CONFIG_SND_SOC=y CONFIG_SND_SOC_INTEL_AVS=y When CONFIG_FORTIFY_SOURCE is enabled, memcmp() (called from guid_equal()) becomes a wrapper to do compile time checking, which interacts poorly with inlining plus CONFIG_KASAN=y. With ARCH=arm64 allmodconfig + CONFIG_KASAN=n + CONFIG_FRAME_WARN=128, the stack usage is much better: sound/soc/intel/avs/path.c:815:18: warning: stack frame size (624) exceeds limit (128) in 'avs_path_create' [-Wframe-larger-than] struct avs_path *avs_path_create(struct avs_dev *adev, u32 dma_id, ^ sound/soc/intel/avs/path.c:873:5: warning: stack frame size (144) exceeds limit (128) in 'avs_path_bind' [-Wframe-larger-than] int avs_path_bind(struct avs_path *path) ^ 2 warnings generated. To avoid this warning, mark avs_path_module_type_create() as noinline_for_stack, which redistributes the stack usage across multiple functions, regardless of CONFIG_KASAN. With ARCH=arm64 allmodconfig + CONFIG_FRAME_WARN=128, the warnings show: avs_path_create(): 192 avs_path_bind(): 272 avs_path_module_type_create(): 416 avs_mux_create(): 160 avs_updown_mix_create(): 160 avs_aec_create(): 176 avs_asrc_create(): 144 With ARCH=arm64 allmodconfig + CONFIG_FRAME_WARN=128 + CONFIG_KASAN=n, the warnings show: avs_path_create(): 192 avs_path_bind(): 144 avs_path_module_type_create(): 416 avs_mux_create(): 176 avs_updown_mix_create(): 176 avs_src_create(): 144 avs_aec_create(): 192 avs_asrc_create(): 144 avs_wov_create(): 144 Link: https://github.com/ClangBuiltLinux/linux/issues/1642 Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/intel/avs/path.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 3d46dd5e5bc4..ec2aa0001f91 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -449,7 +449,8 @@ static int avs_modext_create(struct avs_dev *adev, struct avs_path_module *mod) return ret; } -static int avs_path_module_type_create(struct avs_dev *adev, struct avs_path_module *mod) +static noinline_for_stack int avs_path_module_type_create(struct avs_dev *adev, + struct avs_path_module *mod) { const guid_t *type = &mod->template->cfg_ext->type; base-commit: ff6992735ade75aae3e35d16b17da1008d753d28 -- 2.37.1

3 4

[PATCH v5 2/4] PCI: Move isa_dma_bridge_buggy out of dma.h
by Stafford Horne 21 Jul '22

21 Jul '22

During recent PCI cleanups we noticed that the isa_dma_bridge_buggy symbol supported by all architectures is actually only used for x86_32. This patch moves the symbol out of all architectures limiting usage to only x86_32. This is possible because only x86_32 platforms or quirks existing in PCI devices supported on x86_32 ever set this. A new global header linux/isa-dma.h is added to provide a common place to maintain the definition. Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Suggested-by: Christoph Hellwig <hch(a)infradead.org> Acked-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reviewed-by: Christoph Hellwig <hch(a)lst.de> Signed-off-by: Stafford Horne <shorne(a)gmail.com> --- Since v4: - Also remove isa_dma_bridge_buggy from arm64, csky, riscv as this point. arch/alpha/include/asm/dma.h | 9 --------- arch/arc/include/asm/dma.h | 5 ----- arch/arm/include/asm/dma.h | 6 ------ arch/arm64/include/asm/pci.h | 2 -- arch/csky/include/asm/pci.h | 2 -- arch/ia64/include/asm/dma.h | 2 -- arch/m68k/include/asm/dma.h | 6 ------ arch/microblaze/include/asm/dma.h | 6 ------ arch/mips/include/asm/dma.h | 8 -------- arch/parisc/include/asm/dma.h | 6 ------ arch/powerpc/include/asm/dma.h | 6 ------ arch/riscv/include/asm/pci.h | 2 -- arch/s390/include/asm/dma.h | 6 ------ arch/sh/include/asm/dma.h | 6 ------ arch/sparc/include/asm/dma.h | 8 -------- arch/um/include/asm/pci.h | 2 -- arch/x86/include/asm/dma.h | 8 -------- arch/xtensa/include/asm/dma.h | 7 ------- drivers/comedi/drivers/comedi_isadma.c | 2 +- drivers/pci/pci.c | 2 ++ drivers/pci/quirks.c | 4 +++- include/linux/isa-dma.h | 14 ++++++++++++++ sound/core/isadma.c | 2 +- 23 files changed, 21 insertions(+), 100 deletions(-) create mode 100644 include/linux/isa-dma.h diff --git a/arch/alpha/include/asm/dma.h b/arch/alpha/include/asm/dma.h index 28610ea7786d..a04d76b96089 100644 --- a/arch/alpha/include/asm/dma.h +++ b/arch/alpha/include/asm/dma.h @@ -365,13 +365,4 @@ extern void free_dma(unsigned int dmanr); /* release it again */ #define KERNEL_HAVE_CHECK_DMA extern int check_dma(unsigned int dmanr); -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - - #endif /* _ASM_DMA_H */ diff --git a/arch/arc/include/asm/dma.h b/arch/arc/include/asm/dma.h index 5b744f4b10a7..02431027ed2f 100644 --- a/arch/arc/include/asm/dma.h +++ b/arch/arc/include/asm/dma.h @@ -7,10 +7,5 @@ #define ASM_ARC_DMA_H #define MAX_DMA_ADDRESS 0xC0000000 -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy 0 -#endif #endif diff --git a/arch/arm/include/asm/dma.h b/arch/arm/include/asm/dma.h index a81dda65c576..907d139be431 100644 --- a/arch/arm/include/asm/dma.h +++ b/arch/arm/include/asm/dma.h @@ -143,10 +143,4 @@ extern int get_dma_residue(unsigned int chan); #endif /* CONFIG_ISA_DMA_API */ -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __ASM_ARM_DMA_H */ diff --git a/arch/arm64/include/asm/pci.h b/arch/arm64/include/asm/pci.h index 0aebc3488c32..682c922b5658 100644 --- a/arch/arm64/include/asm/pci.h +++ b/arch/arm64/include/asm/pci.h @@ -20,8 +20,6 @@ #define arch_can_pci_mmap_wc() 1 #define ARCH_GENERIC_PCI_MMAP_RESOURCE 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/csky/include/asm/pci.h b/arch/csky/include/asm/pci.h index 0535f1aaae38..5c02454ec724 100644 --- a/arch/csky/include/asm/pci.h +++ b/arch/csky/include/asm/pci.h @@ -15,8 +15,6 @@ /* C-SKY shim does not initialize PCI bus */ #define pcibios_assign_all_busses() 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/ia64/include/asm/dma.h b/arch/ia64/include/asm/dma.h index 59625e9c1f9c..eaed2626ffda 100644 --- a/arch/ia64/include/asm/dma.h +++ b/arch/ia64/include/asm/dma.h @@ -12,8 +12,6 @@ extern unsigned long MAX_DMA_ADDRESS; -extern int isa_dma_bridge_buggy; - #define free_dma(x) #endif /* _ASM_IA64_DMA_H */ diff --git a/arch/m68k/include/asm/dma.h b/arch/m68k/include/asm/dma.h index f6c5e0dfb4e5..1c8d9c5bc2fa 100644 --- a/arch/m68k/include/asm/dma.h +++ b/arch/m68k/include/asm/dma.h @@ -6,10 +6,4 @@ bootmem allocator (but this should do it for this) */ #define MAX_DMA_ADDRESS PAGE_OFFSET -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _M68K_DMA_H */ diff --git a/arch/microblaze/include/asm/dma.h b/arch/microblaze/include/asm/dma.h index f801582be912..7484c9eb66c4 100644 --- a/arch/microblaze/include/asm/dma.h +++ b/arch/microblaze/include/asm/dma.h @@ -9,10 +9,4 @@ /* Virtual address corresponding to last available physical memory address. */ #define MAX_DMA_ADDRESS (CONFIG_KERNEL_START + memory_size - 1) -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_MICROBLAZE_DMA_H */ diff --git a/arch/mips/include/asm/dma.h b/arch/mips/include/asm/dma.h index be726b943530..d6186e6bea7e 100644 --- a/arch/mips/include/asm/dma.h +++ b/arch/mips/include/asm/dma.h @@ -307,12 +307,4 @@ static __inline__ int get_dma_residue(unsigned int dmanr) extern int request_dma(unsigned int dmanr, const char * device_id); /* reserve a DMA channel */ extern void free_dma(unsigned int dmanr); /* release it again */ -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_DMA_H */ diff --git a/arch/parisc/include/asm/dma.h b/arch/parisc/include/asm/dma.h index eea80ed34e6d..9e8c101de902 100644 --- a/arch/parisc/include/asm/dma.h +++ b/arch/parisc/include/asm/dma.h @@ -176,10 +176,4 @@ static __inline__ void set_dma_count(unsigned int dmanr, unsigned int count) #define free_dma(dmanr) -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_DMA_H */ diff --git a/arch/powerpc/include/asm/dma.h b/arch/powerpc/include/asm/dma.h index 6161a9596196..d97c66d9ae34 100644 --- a/arch/powerpc/include/asm/dma.h +++ b/arch/powerpc/include/asm/dma.h @@ -340,11 +340,5 @@ extern int request_dma(unsigned int dmanr, const char *device_id); /* release it again */ extern void free_dma(unsigned int dmanr); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __KERNEL__ */ #endif /* _ASM_POWERPC_DMA_H */ diff --git a/arch/riscv/include/asm/pci.h b/arch/riscv/include/asm/pci.h index a7b8f0d0df7f..f904df586c03 100644 --- a/arch/riscv/include/asm/pci.h +++ b/arch/riscv/include/asm/pci.h @@ -20,8 +20,6 @@ #define ARCH_GENERIC_PCI_MMAP_RESOURCE 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/s390/include/asm/dma.h b/arch/s390/include/asm/dma.h index 6f26f35d4a71..dec1c4ce628c 100644 --- a/arch/s390/include/asm/dma.h +++ b/arch/s390/include/asm/dma.h @@ -11,10 +11,4 @@ */ #define MAX_DMA_ADDRESS 0x80000000 -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_S390_DMA_H */ diff --git a/arch/sh/include/asm/dma.h b/arch/sh/include/asm/dma.h index 17d23ae98c77..c8bee3f985a2 100644 --- a/arch/sh/include/asm/dma.h +++ b/arch/sh/include/asm/dma.h @@ -137,10 +137,4 @@ extern int register_chan_caps(const char *dmac, struct dma_chan_caps *capslist); extern int dma_create_sysfs_files(struct dma_channel *, struct dma_info *); extern void dma_remove_sysfs_files(struct dma_channel *, struct dma_info *); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __ASM_SH_DMA_H */ diff --git a/arch/sparc/include/asm/dma.h b/arch/sparc/include/asm/dma.h index 462e7c794a09..08043f35b110 100644 --- a/arch/sparc/include/asm/dma.h +++ b/arch/sparc/include/asm/dma.h @@ -82,14 +82,6 @@ #define DMA_BURST64 0x40 #define DMA_BURSTBITS 0x7f -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #ifdef CONFIG_SPARC32 struct device; diff --git a/arch/um/include/asm/pci.h b/arch/um/include/asm/pci.h index 26b96c02ef61..1211855aff34 100644 --- a/arch/um/include/asm/pci.h +++ b/arch/um/include/asm/pci.h @@ -9,8 +9,6 @@ #define pcibios_assign_all_busses() 1 -extern int isa_dma_bridge_buggy; - #ifdef CONFIG_PCI_DOMAINS static inline int pci_proc_domain(struct pci_bus *bus) { diff --git a/arch/x86/include/asm/dma.h b/arch/x86/include/asm/dma.h index 8e95aa4b0d17..8ae6e0e11b8b 100644 --- a/arch/x86/include/asm/dma.h +++ b/arch/x86/include/asm/dma.h @@ -307,12 +307,4 @@ extern int request_dma(unsigned int dmanr, const char *device_id); extern void free_dma(unsigned int dmanr); #endif -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_X86_DMA_H */ diff --git a/arch/xtensa/include/asm/dma.h b/arch/xtensa/include/asm/dma.h index bb099a373b5a..172644539032 100644 --- a/arch/xtensa/include/asm/dma.h +++ b/arch/xtensa/include/asm/dma.h @@ -52,11 +52,4 @@ extern int request_dma(unsigned int dmanr, const char * device_id); extern void free_dma(unsigned int dmanr); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - - #endif diff --git a/drivers/comedi/drivers/comedi_isadma.c b/drivers/comedi/drivers/comedi_isadma.c index 700982464c53..020b3d1e1ac0 100644 --- a/drivers/comedi/drivers/comedi_isadma.c +++ b/drivers/comedi/drivers/comedi_isadma.c @@ -8,7 +8,7 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/dma-mapping.h> -#include <asm/dma.h> +#include <linux/isa-dma.h> #include <linux/comedi/comedidev.h> #include <linux/comedi/comedi_isadma.h> diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index cfaf40a540a8..60c55d2cb2cc 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -41,8 +41,10 @@ const char *pci_power_names[] = { }; EXPORT_SYMBOL_GPL(pci_power_names); +#ifdef CONFIG_X86_32 int isa_dma_bridge_buggy; EXPORT_SYMBOL(isa_dma_bridge_buggy); +#endif int pci_pci_problems; EXPORT_SYMBOL(pci_pci_problems); diff --git a/drivers/pci/quirks.c b/drivers/pci/quirks.c index 41aeaa235132..6fc64509eee7 100644 --- a/drivers/pci/quirks.c +++ b/drivers/pci/quirks.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/export.h> #include <linux/pci.h> +#include <linux/isa-dma.h> /* isa_dma_bridge_buggy */ #include <linux/init.h> #include <linux/delay.h> #include <linux/acpi.h> @@ -30,7 +31,6 @@ #include <linux/pm_runtime.h> #include <linux/suspend.h> #include <linux/switchtec.h> -#include <asm/dma.h> /* isa_dma_bridge_buggy */ #include "pci.h" static ktime_t fixup_debug_start(struct pci_dev *dev, @@ -239,6 +239,7 @@ static void quirk_passive_release(struct pci_dev *dev) DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82441, quirk_passive_release); DECLARE_PCI_FIXUP_RESUME(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82441, quirk_passive_release); +#ifdef CONFIG_X86_32 /* * The VIA VP2/VP3/MVP3 seem to have some 'features'. There may be a * workaround but VIA don't answer queries. If you happen to have good @@ -265,6 +266,7 @@ DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M1533, quirk_isa_dma DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_1, quirk_isa_dma_hangs); DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_2, quirk_isa_dma_hangs); DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_3, quirk_isa_dma_hangs); +#endif /* * Intel NM10 "TigerPoint" LPC PM1a_STS.BM_STS must be clear diff --git a/include/linux/isa-dma.h b/include/linux/isa-dma.h new file mode 100644 index 000000000000..61504a8c1b9e --- /dev/null +++ b/include/linux/isa-dma.h @@ -0,0 +1,14 @@ +/* SPDX-License-Identifier: GPL-2.0 */ + +#ifndef __LINUX_ISA_DMA_H +#define __LINUX_ISA_DMA_H + +#include <asm/dma.h> + +#if defined(CONFIG_PCI) && defined(CONFIG_X86_32) +extern int isa_dma_bridge_buggy; +#else +#define isa_dma_bridge_buggy (0) +#endif + +#endif /* __LINUX_ISA_DMA_H */ diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 1f45ede023b4..18a86212e3a8 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -12,8 +12,8 @@ #undef HAVE_REALLY_SLOW_DMA_CONTROLLER #include <linux/export.h> +#include <linux/isa-dma.h> #include <sound/core.h> -#include <asm/dma.h> /** * snd_dma_program - program an ISA DMA transfer -- 2.36.1

1 0

[PATCH] docs: driver-api: firmware: add driver firmware guidelines. (v2)
by Dave Airlie 21 Jul '22

21 Jul '22

From: Dave Airlie <airlied(a)redhat.com> A recent snafu where Intel ignored upstream feedback on a firmware change, led to a late rc6 fix being required. In order to avoid this in the future we should document some expectations around linux-firmware. I was originally going to write this for drm, but it seems quite generic advice. v2: rewritten with suggestions from Thorsten Leemhuis. Acked-by: Luis Chamberlain <mcgrof(a)kernel.org> Acked-by: Rodrigo Vivi <rodrigo.vivi(a)intel.com> Signed-off-by: Dave Airlie <airlied(a)redhat.com> --- Documentation/driver-api/firmware/core.rst | 1 + .../firmware/firmware-usage-guidelines.rst | 34 +++++++++++++++++++ 2 files changed, 35 insertions(+) create mode 100644 Documentation/driver-api/firmware/firmware-usage-guidelines.rst diff --git a/Documentation/driver-api/firmware/core.rst b/Documentation/driver-api/firmware/core.rst index 1d1688cbc078..803cd574bbd7 100644 --- a/Documentation/driver-api/firmware/core.rst +++ b/Documentation/driver-api/firmware/core.rst @@ -13,4 +13,5 @@ documents these features. direct-fs-lookup fallback-mechanisms lookup-order + firmware-usage-guidelines diff --git a/Documentation/driver-api/firmware/firmware-usage-guidelines.rst b/Documentation/driver-api/firmware/firmware-usage-guidelines.rst new file mode 100644 index 000000000000..34d2412e78c6 --- /dev/null +++ b/Documentation/driver-api/firmware/firmware-usage-guidelines.rst @@ -0,0 +1,34 @@ +=================== +Firmware Guidelines +=================== + +Drivers that use firmware from linux-firmware should attempt to follow +the rules in this guide. + +* Firmware should be versioned with at least a major/minor version. It + is suggested that the firmware files in linux-firmware be named with + some device specific name, and just the major version. The + major/minor/patch versions should be stored in a header in the + firmware file for the driver to detect any non-ABI fixes/issues. The + firmware files in linux-firmware should be overwritten with the newest + compatible major version. Newer major version firmware should remain + compatible with all kernels that load that major number. + +* Users should *not* have to install newer firmware to use existing + hardware when they install a newer kernel. If the hardware isn't + enabled by default or under development, this can be ignored, until + the first kernel release that enables that hardware. This means no + major version bumps without the kernel retaining backwards + compatibility for the older major versions. Minor version bumps + should not introduce new features that newer kernels depend on + non-optionally. + +* If a security fix needs lockstep firmware and kernel fixes in order to + be successful, then all supported major versions in the linux-firmware + repo should be updated with the security fix, and the kernel patches + should detect if the firmware is new enough to declare if the security + issue is fixed. All communications around security fixes should point + at both the firmware and kernel fixes. If a security fix requires + deprecating old major versions, then this should only be done as a + last option, and be stated clearly in all communications. + -- 2.36.1

3 3

[PATCH v2 1/2] dt-bindings: dsp: fsl: Add SOF compatile string for i.MX8ULP
by Shengjiu Wang 21 Jul '22

21 Jul '22

Add SOF compatile string "fsl,imx8ulp-dsp" for supporting DSP device on i.MX8ULP platform. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Rob Herring <robh(a)kernel.org> --- changes in v2 - add acked-by Rob Documentation/devicetree/bindings/dsp/fsl,dsp.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml index e66ef2da7879..9af40da5688e 100644 --- a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx8qxp-dsp - fsl,imx8qm-dsp - fsl,imx8mp-dsp + - fsl,imx8ulp-dsp - fsl,imx8qxp-hifi4 - fsl,imx8qm-hifi4 - fsl,imx8mp-hifi4 -- 2.34.1

1 1

[PATCH v10 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 21 Jul '22

21 Jul '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v9: - Drop the definitions that unused in mt8186-afe-common.h. - Use device_get_match_data() instead of of_device_get_match_data(). Changes since v8: - Remove unnecessary comments and blank line. - Use devm_add_action_or_reset to ease error handling when dev probe. - Remove .remove callback in plaform driver. Changes since v7: - Optimize the code logic, such as the return value and position of a function. - Use devm_pm_runtime_enable() instead of pm_runtime_enable(). - Use pm_runtime_resume_and_get() instead of pm_runtime_get_sync(). Changes since v6: - The Makefile and Kconfig updates are moved into the driver patches so that can keep independence of each patch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 2 +- sound/soc/mediatek/mt8186/mt8186-afe-common.h | 195 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 255 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3000 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 57 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 16 files changed, 5902 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.18.0

2 9

[PATCH] ASoC: Makefile: Fix simultaneous build of KUNIT tests
by Amadeusz Sławiński 21 Jul '22

21 Jul '22

Using obj-$() := instead of obj-$() += leads to the latter assignment overwriting earlier value. Fix this by using incremental assignment to add additional objects to build. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d4528962ac34..453181ef6c94 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -9,12 +9,12 @@ endif ifneq ($(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST),) # snd-soc-test-objs := soc-topology-test.o -obj-$(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST) := soc-topology-test.o +obj-$(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST) += soc-topology-test.o endif ifneq ($(CONFIG_SND_SOC_UTILS_KUNIT_TEST),) # snd-soc-test-objs := soc-utils-test.o -obj-$(CONFIG_SND_SOC_UTILS_KUNIT_TEST) := soc-utils-test.o +obj-$(CONFIG_SND_SOC_UTILS_KUNIT_TEST) += soc-utils-test.o endif ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) -- 2.25.1

2 1

[PATCH v5] update tas2780.yaml
by Raphael-Xu 21 Jul '22

21 Jul '22

1.remove unused item in DT 2.revise the format Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../devicetree/bindings/sound/tas2780.yaml | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2780.yaml b/Documentation/devicetree/bindings/sound/tas2780.yaml index 5a0de0995066..21389a16fdb9 100644 --- a/Documentation/devicetree/bindings/sound/tas2780.yaml +++ b/Documentation/devicetree/bindings/sound/tas2780.yaml @@ -22,24 +22,29 @@ properties: - ti,tas2780 reg: - maxItems: 1 - description: | - I2C address of the device can be between 0x38 to 0x45. + description: + I2C address of the device can be between 0x38 to 0x3f. + minimum: 0x38 + maximum: 0x3f reset-gpios: - maxItems: 1 description: GPIO used to reset the device. + maxItems: 1 interrupts: maxItems: 1 ti,imon-slot-no: - $ref: /schemas/types.yaml#/definitions/uint32 description: TDM TX current sense time slot. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 ti,vmon-slot-no: - $ref: /schemas/types.yaml#/definitions/uint32 description: TDM TX voltage sense time slot. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 '#sound-dai-cells': const: 1 @@ -60,10 +65,8 @@ examples: compatible = "ti,tas2780"; reg = <0x38>; #sound-dai-cells = <1>; - interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; - shutdown-gpios = <&gpio1 15 0>; ti,imon-slot-no = <0>; ti,vmon-slot-no = <2>; }; -- 2.35.1

3 4

[PATCH 1/2] ASoC: SOF: imx: Add i.MX8ULP HW support
by Shengjiu Wang 21 Jul '22

21 Jul '22

From: Zhang Peng <peng.zhang_8(a)nxp.com> This adds skeleton support for the audio DSP hardware found on NXP i.MX8ULP platform. On i.MX8ULP resources (clocks, power, etc) are managed by the System Integration Module in LPAV domain and XRDC which is handled by arm trusted firmware. Signed-off-by: Zhang Peng <peng.zhang_8(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/sof/imx/Kconfig | 9 + sound/soc/sof/imx/Makefile | 2 + sound/soc/sof/imx/imx8ulp.c | 541 ++++++++++++++++++++++++++++++++++++ 3 files changed, 552 insertions(+) create mode 100644 sound/soc/sof/imx/imx8ulp.c diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index cc6e695f913a..4751b04d5e6f 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -41,4 +41,13 @@ config SND_SOC_SOF_IMX8M Say Y if you have such a device. If unsure select "N". +config SND_SOC_SOF_IMX8ULP + tristate "SOF support for i.MX8ULP" + depends on IMX_DSP + select SND_SOC_SOF_IMX_COMMON + help + This adds support for Sound Open Firmware for NXP i.MX8ULP platforms. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_IMX_TOPLEVEL diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile index dba93c3466ec..798b43a415bf 100644 --- a/sound/soc/sof/imx/Makefile +++ b/sound/soc/sof/imx/Makefile @@ -1,9 +1,11 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) snd-sof-imx8-objs := imx8.o snd-sof-imx8m-objs := imx8m.o +snd-sof-imx8ulp-objs := imx8ulp.o snd-sof-imx-common-objs := imx-common.o obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o obj-$(CONFIG_SND_SOC_SOF_IMX8M) += snd-sof-imx8m.o +obj-$(CONFIG_SND_SOC_SOF_IMX8ULP) += snd-sof-imx8ulp.o obj-$(CONFIG_SND_SOC_SOF_IMX_COMMON) += imx-common.o diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c new file mode 100644 index 000000000000..ecb8750de7ea --- /dev/null +++ b/sound/soc/sof/imx/imx8ulp.c @@ -0,0 +1,541 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// Copyright 2021-2022 NXP +// +// Author: Peng Zhang <peng.zhang_8(a)nxp.com> +// +// Hardware interface for audio DSP on i.MX8ULP + +#include <linux/clk.h> +#include <linux/firmware.h> +#include <linux/firmware/imx/dsp.h> +#include <linux/firmware/imx/ipc.h> +#include <linux/firmware/imx/svc/misc.h> +#include <linux/mfd/syscon.h> +#include <linux/module.h> +#include <linux/of_address.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <linux/of_reserved_mem.h> + +#include <sound/sof.h> +#include <sound/sof/xtensa.h> + +#include "../ops.h" +#include "../sof-of-dev.h" +#include "imx-common.h" + +#define FSL_SIP_HIFI_XRDC 0xc200000e + +/* SIM Domain register */ +#define SYSCTRL0 0x8 +#define EXECUTE_BIT BIT(13) +#define RESET_BIT BIT(16) +#define HIFI4_CLK_BIT BIT(17) +#define PB_CLK_BIT BIT(18) +#define PLAT_CLK_BIT BIT(19) +#define DEBUG_LOGIC_BIT BIT(25) + +#define MBOX_OFFSET 0x800000 +#define MBOX_SIZE 0x1000 + +struct arm_smccc_res smc_resource; + +static struct clk_bulk_data imx8ulp_dsp_clks[] = { + { .id = "core" }, + { .id = "ipg" }, + { .id = "ocram" }, + { .id = "mu" }, +}; + +struct imx8ulp_priv { + struct device *dev; + struct snd_sof_dev *sdev; + + /* DSP IPC handler */ + struct imx_dsp_ipc *dsp_ipc; + struct platform_device *ipc_dev; + + struct regmap *regmap; + struct imx_clocks *clks; +}; + +static void imx8ulp_sim_lpav_start(struct imx8ulp_priv *priv) +{ + /* Controls the HiFi4 DSP Reset: 1 in reset, 0 out of reset */ + regmap_update_bits(priv->regmap, SYSCTRL0, RESET_BIT, 0); + /* Reset HiFi4 DSP Debug logic: 1 reset, 0 not set */ + regmap_update_bits(priv->regmap, SYSCTRL0, DEBUG_LOGIC_BIT, 0); + /* Stall HIFI4 DSP Execution: 1 stall, 0 not stall */ + regmap_update_bits(priv->regmap, SYSCTRL0, EXECUTE_BIT, 0); +} + +static void imx8ulp_get_reply(struct snd_sof_dev *sdev) +{ + struct snd_sof_ipc_msg *msg = sdev->msg; + struct sof_ipc_reply reply; + int ret = 0; + + if (!msg) { + dev_warn(sdev->dev, "unexpected ipc interrupt\n"); + return; + } + + /* get reply */ + sof_mailbox_read(sdev, sdev->host_box.offset, &reply, sizeof(reply)); + + if (reply.error < 0) { + memcpy(msg->reply_data, &reply, sizeof(reply)); + ret = reply.error; + } else { + /* reply has correct size? */ + if (reply.hdr.size != msg->reply_size) { + dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", + msg->reply_size, reply.hdr.size); + ret = -EINVAL; + } + + /* read the message */ + if (msg->reply_size > 0) + sof_mailbox_read(sdev, sdev->host_box.offset, + msg->reply_data, msg->reply_size); + } + + msg->reply_error = ret; +} + +static int imx8ulp_get_mailbox_offset(struct snd_sof_dev *sdev) +{ + return MBOX_OFFSET; +} + +static int imx8ulp_get_window_offset(struct snd_sof_dev *sdev, u32 id) +{ + return MBOX_OFFSET; +} + +static void imx8ulp_dsp_handle_reply(struct imx_dsp_ipc *ipc) +{ + struct imx8ulp_priv *priv = imx_dsp_get_data(ipc); + unsigned long flags; + + spin_lock_irqsave(&priv->sdev->ipc_lock, flags); + + imx8ulp_get_reply(priv->sdev); + snd_sof_ipc_reply(priv->sdev, 0); + spin_unlock_irqrestore(&priv->sdev->ipc_lock, flags); +} + +static void imx8ulp_dsp_handle_request(struct imx_dsp_ipc *ipc) +{ + struct imx8ulp_priv *priv = imx_dsp_get_data(ipc); + u32 p; /* panic code */ + + /* Read the message from the debug box. */ + sof_mailbox_read(priv->sdev, priv->sdev->debug_box.offset + 4, &p, sizeof(p)); + + /* Check to see if the message is a panic code (0x0dead***) */ + if ((p & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) + snd_sof_dsp_panic(priv->sdev, p, true); + else + snd_sof_ipc_msgs_rx(priv->sdev); +} + +static struct imx_dsp_ops dsp_ops = { + .handle_reply = imx8ulp_dsp_handle_reply, + .handle_request = imx8ulp_dsp_handle_request, +}; + +static int imx8ulp_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + imx_dsp_ring_doorbell(priv->dsp_ipc, 0); + + return 0; +} + +static int imx8ulp_run(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + + imx8ulp_sim_lpav_start(priv); + + return 0; +} + +static int imx8ulp_reset(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + + /* HiFi4 Platform Clock Enable: 1 enabled, 0 disabled */ + regmap_update_bits(priv->regmap, SYSCTRL0, PLAT_CLK_BIT, PLAT_CLK_BIT); + /* HiFi4 PBCLK clock enable: 1 enabled, 0 disabled */ + regmap_update_bits(priv->regmap, SYSCTRL0, PB_CLK_BIT, PB_CLK_BIT); + /* HiFi4 Clock Enable: 1 enabled, 0 disabled */ + regmap_update_bits(priv->regmap, SYSCTRL0, HIFI4_CLK_BIT, HIFI4_CLK_BIT); + + regmap_update_bits(priv->regmap, SYSCTRL0, RESET_BIT, RESET_BIT); + usleep_range(1, 2); + /* Stall HIFI4 DSP Execution: 1 stall, 0 not stall */ + regmap_update_bits(priv->regmap, SYSCTRL0, EXECUTE_BIT, EXECUTE_BIT); + usleep_range(1, 2); + + arm_smccc_smc(FSL_SIP_HIFI_XRDC, 0, 0, 0, 0, 0, 0, 0, &smc_resource); + + return 0; +} + +static int imx8ulp_probe(struct snd_sof_dev *sdev) +{ + struct platform_device *pdev = + container_of(sdev->dev, struct platform_device, dev); + struct device_node *np = pdev->dev.of_node; + struct device_node *res_node; + struct resource *mmio; + struct imx8ulp_priv *priv; + struct resource res; + u32 base, size; + int ret = 0; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL); + if (!priv->clks) + return -ENOMEM; + + sdev->num_cores = 1; + sdev->pdata->hw_pdata = priv; + priv->dev = sdev->dev; + priv->sdev = sdev; + + /* System integration module(SIM) control dsp configurtion */ + priv->regmap = syscon_regmap_lookup_by_phandle(np, "fsl,dsp-ctrl"); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + priv->ipc_dev = platform_device_register_data(sdev->dev, "imx-dsp", + PLATFORM_DEVID_NONE, + pdev, sizeof(*pdev)); + if (IS_ERR(priv->ipc_dev)) + return PTR_ERR(priv->ipc_dev); + + priv->dsp_ipc = dev_get_drvdata(&priv->ipc_dev->dev); + if (!priv->dsp_ipc) { + /* DSP IPC driver not probed yet, try later */ + ret = -EPROBE_DEFER; + dev_err(sdev->dev, "Failed to get drvdata\n"); + goto exit_pdev_unregister; + } + + imx_dsp_set_data(priv->dsp_ipc, priv); + priv->dsp_ipc->ops = &dsp_ops; + + /* DSP base */ + mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mmio) { + base = mmio->start; + size = resource_size(mmio); + } else { + dev_err(sdev->dev, "error: failed to get DSP base at idx 0\n"); + ret = -EINVAL; + goto exit_pdev_unregister; + } + + sdev->bar[SOF_FW_BLK_TYPE_IRAM] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[SOF_FW_BLK_TYPE_IRAM]) { + dev_err(sdev->dev, "failed to ioremap base 0x%x size 0x%x\n", + base, size); + ret = -ENODEV; + goto exit_pdev_unregister; + } + sdev->mmio_bar = SOF_FW_BLK_TYPE_IRAM; + + res_node = of_parse_phandle(np, "memory-reserved", 0); + if (!res_node) { + dev_err(&pdev->dev, "failed to get memory region node\n"); + ret = -ENODEV; + goto exit_pdev_unregister; + } + + ret = of_address_to_resource(res_node, 0, &res); + if (ret) { + dev_err(&pdev->dev, "failed to get reserved region address\n"); + goto exit_pdev_unregister; + } + + sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, res.start, + resource_size(&res)); + if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { + dev_err(sdev->dev, "failed to ioremap mem 0x%x size 0x%x\n", + base, size); + ret = -ENOMEM; + goto exit_pdev_unregister; + } + sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM; + + /* set default mailbox offset for FW ready message */ + sdev->dsp_box.offset = MBOX_OFFSET; + + ret = of_reserved_mem_device_init(sdev->dev); + if (ret) { + dev_err(&pdev->dev, "failed to init reserved memory region %d\n", ret); + goto exit_pdev_unregister; + } + + priv->clks->dsp_clks = imx8ulp_dsp_clks; + priv->clks->num_dsp_clks = ARRAY_SIZE(imx8ulp_dsp_clks); + + ret = imx8_parse_clocks(sdev, priv->clks); + if (ret < 0) + goto exit_pdev_unregister; + + ret = imx8_enable_clocks(sdev, priv->clks); + if (ret < 0) + goto exit_pdev_unregister; + + return 0; + +exit_pdev_unregister: + platform_device_unregister(priv->ipc_dev); + + return ret; +} + +static int imx8ulp_remove(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; + + platform_device_unregister(priv->ipc_dev); + + return 0; +} + +/* on i.MX8 there is 1 to 1 match between type and BAR idx */ +static int imx8ulp_get_bar_index(struct snd_sof_dev *sdev, u32 type) +{ + return type; +} + +static int imx8ulp_suspend(struct snd_sof_dev *sdev) +{ + int i; + struct imx8ulp_priv *priv = (struct imx8ulp_priv *)sdev->pdata->hw_pdata; + + regmap_update_bits(priv->regmap, SYSCTRL0, EXECUTE_BIT, EXECUTE_BIT); + + for (i = 0; i < DSP_MU_CHAN_NUM; i++) + imx_dsp_free_channel(priv->dsp_ipc, i); + + imx8_disable_clocks(sdev, priv->clks); + + return 0; +} + +static int imx8ulp_resume(struct snd_sof_dev *sdev) +{ + struct imx8ulp_priv *priv = (struct imx8ulp_priv *)sdev->pdata->hw_pdata; + int i; + + imx8_enable_clocks(sdev, priv->clks); + + for (i = 0; i < DSP_MU_CHAN_NUM; i++) + imx_dsp_request_channel(priv->dsp_ipc, i); + + return 0; +} + +static int imx8ulp_dsp_runtime_resume(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = SOF_DSP_PM_D0, + .substate = 0, + }; + + imx8ulp_resume(sdev); + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static int imx8ulp_dsp_runtime_suspend(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = SOF_DSP_PM_D3, + .substate = 0, + }; + + imx8ulp_suspend(sdev); + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static int imx8ulp_dsp_suspend(struct snd_sof_dev *sdev, unsigned int target_state) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = target_state, + .substate = 0, + }; + + if (!pm_runtime_suspended(sdev->dev)) + imx8ulp_suspend(sdev); + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static int imx8ulp_dsp_resume(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = SOF_DSP_PM_D0, + .substate = 0, + }; + + imx8ulp_resume(sdev); + + if (pm_runtime_suspended(sdev->dev)) { + pm_runtime_disable(sdev->dev); + pm_runtime_set_active(sdev->dev); + pm_runtime_mark_last_busy(sdev->dev); + pm_runtime_enable(sdev->dev); + pm_runtime_idle(sdev->dev); + } + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static struct snd_soc_dai_driver imx8ulp_dai[] = { + { + .name = "sai5", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, + }, + { + .name = "sai6", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, + }, +}; + +static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) +{ + sdev->dsp_power_state = *target_state; + + return 0; +} + +/* i.MX8 ops */ +struct snd_sof_dsp_ops sof_imx8ulp_ops = { + /* probe and remove */ + .probe = imx8ulp_probe, + .remove = imx8ulp_remove, + /* DSP core boot */ + .run = imx8ulp_run, + .reset = imx8ulp_reset, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* Module IO */ + .read64 = sof_io_read64, + + /* Mailbox IO */ + .mailbox_read = sof_mailbox_read, + .mailbox_write = sof_mailbox_write, + + /* ipc */ + .send_msg = imx8ulp_send_msg, + .get_mailbox_offset = imx8ulp_get_mailbox_offset, + .get_window_offset = imx8ulp_get_window_offset, + + .ipc_msg_data = sof_ipc_msg_data, + .set_stream_data_offset = sof_set_stream_data_offset, + + /* stream callbacks */ + .pcm_open = sof_stream_pcm_open, + .pcm_close = sof_stream_pcm_close, + + /* module loading */ + .get_bar_index = imx8ulp_get_bar_index, + /* firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* Debug information */ + .dbg_dump = imx8_dump, + + /* Firmware ops */ + .dsp_arch_ops = &sof_xtensa_arch_ops, + + /* DAI drivers */ + .drv = imx8ulp_dai, + .num_drv = ARRAY_SIZE(imx8ulp_dai), + + /* ALSA HW info flags */ + .hw_info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + + /* PM */ + .runtime_suspend = imx8ulp_dsp_runtime_suspend, + .runtime_resume = imx8ulp_dsp_runtime_resume, + + .suspend = imx8ulp_dsp_suspend, + .resume = imx8ulp_dsp_resume, + + .set_power_state = imx8ulp_dsp_set_power_state, +}; + +static struct sof_dev_desc sof_of_imx8ulp_desc = { + .ipc_supported_mask = BIT(SOF_IPC), + .ipc_default = SOF_IPC, + .default_fw_path = { + [SOF_IPC] = "imx/sof", + }, + .default_tplg_path = { + [SOF_IPC] = "imx/sof-tplg", + }, + .default_fw_filename = { + [SOF_IPC] = "sof-imx8ulp.ri", + }, + .nocodec_tplg_filename = "sof-imx8ulp-nocodec.tplg", + .ops = &sof_imx8ulp_ops, +}; + +static const struct of_device_id sof_of_imx8ulp_ids[] = { + { .compatible = "fsl,imx8ulp-dsp", .data = &sof_of_imx8ulp_desc}, + { } +}; +MODULE_DEVICE_TABLE(of, sof_of_imx8ulp_ids); + +/* DT driver definition */ +static struct platform_driver snd_sof_of_imx8ulp_driver = { + .probe = sof_of_probe, + .remove = sof_of_remove, + .driver = { + .name = "sof-audio-of-imx8ulp", + .pm = &sof_of_pm, + .of_match_table = sof_of_imx8ulp_ids, + }, +}; +module_platform_driver(snd_sof_of_imx8ulp_driver); + +MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); +MODULE_LICENSE("Dual BSD/GPL"); -- 2.34.1

6 9

[PATCH v1 1/2] ASoC: codecs: add support for ES8326
by Zhu Ning 21 Jul '22

21 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ----- v1 Changes: 1.remove 24kHz support 2.disables micbias during suspend and remove 3.regmap resync during resume 4.make irq optional 5.cancel_delay_work during suspend and remove Unchanged: 1.dts about reg58,59. We ask OEMs to use a specific board layout for irq and jack detection. a dts about it can be added, but only useful in rare situation. --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 806 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 +++++++++ 4 files changed, 1000 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..4c9e844d4c47 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,806 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS_1B, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS_1B, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DETECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + es8326_disable_micbias(comp); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if (!es8326->jack) + goto out; + + es8326_enable_micbias(comp); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8326->regmap, false); + regcache_sync(es8326->regmap); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + es8326_disable_micbias(component); + cancel_delayed_work_sync(&es8326->jack_detect_work); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_OFF); + regcache_cache_only(es8326->regmap, true); + regcache_mark_dirty(es8326->regmap); + + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) { + es8326_disable_micbias(component); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + } + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) { + dev_warn(&i2c->dev, "Failed to reguest IRQ: %d: %d\n", + es8326->irq, ret); + es8326->irq = -ENXIO; + } + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..717f3fd42372 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.h -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

3 3

[PATCH] ASoC:do not initialise statics to 0.
by Xin Gao 20 Jul '22

20 Jul '22

do not initialise statics to 0. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 45c301ea5e00..8a8b25589c55 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -50,7 +50,7 @@ struct cht_mc_private { #define CHT_RT5645_SSP0_AIF2 BIT(18) #define CHT_RT5645_PMC_PLT_CLK_0 BIT(19) -static unsigned long cht_rt5645_quirk = 0; +static unsigned long cht_rt5645_quirk; static void log_quirks(struct device *dev) { -- 2.30.2

2 1

[PATCH] ASoC: dt-bindings: sgtl5000: Add missing type to 'micbias-voltage-m-volts'
by Rob Herring 20 Jul '22

20 Jul '22

'micbias-voltage-m-volts' is missing a type definition. '-m-volts' is not a standard unit (should be '-microvolt'). As the property is already in use, add a type reference. Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/sgtl5000.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml index e762c320b574..2bc7f00ce4a2 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.yaml +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -47,6 +47,7 @@ properties: description: The bias voltage to be used in mVolts. The voltage can take values from 1.25V to 3V by 250mV steps. If this node is not mentioned or the value is unknown, then the value is set to 1.25V. + $ref: "/schemas/types.yaml#/definitions/uint32" enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ] lrclk-strength: -- 2.34.1

2 1

[PATCH v3 0/2] ASoC: dt-bindings: atmel-classd: Convert to
by Ryan.Wanner＠microchip.com 20 Jul '22

20 Jul '22

From: Ryan Wanner <Ryan.Wanner(a)microchip.com> This patch series converts atmel-classd and atmel-pdmic device tree bindings to json-schema. v1 -> v2: - Fix commit formatting. - Fix titles in yaml file. - Removed trivial descriptions. - Correct formatting errors. v2 -> v3: - Fix subsystem prefix. - Fix word wrap formatting. - Removed assigned-clocks parameter. - Correct descriptions in inconsistent locations. - Remove duplicated description. - Correct formatting for dts example. Ryan Wanner (2): ASoC: dt-bindings: atmel-classd: Convert to json-schema ASoC: dt-bindings: atmel-classd: PDMIC convert to json-schema .../bindings/sound/atmel,sama5d2-classd.yaml | 100 ++++++++++++++++++ .../bindings/sound/atmel,sama5d2-pdmic.yaml | 98 +++++++++++++++++ .../bindings/sound/atmel-classd.txt | 55 ---------- .../devicetree/bindings/sound/atmel-pdmic.txt | 55 ---------- 4 files changed, 198 insertions(+), 110 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt delete mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt -- 2.34.1

3 5

[PATCH v4 2/3] PCI: Move isa_dma_bridge_buggy out of dma.h
by Stafford Horne 20 Jul '22

20 Jul '22

During recent PCI cleanups we noticed that the isa_dma_bridge_buggy symbol supported by all architectures is actually only used for x86_32. This patch moves the symbol out of all architectures limiting usage to only x86_32. This is possible because only x86_32 platforms or quirks existing in PCI devices supported on x86_32 ever set this. A new global header linux/isa-dma.h is added to provide a common place to maintain the definition. Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Suggested-by: Christoph Hellwig <hch(a)infradead.org> Signed-off-by: Stafford Horne <shorne(a)gmail.com> --- Since v3: - New patch. arch/alpha/include/asm/dma.h | 9 --------- arch/arc/include/asm/dma.h | 5 ----- arch/arm/include/asm/dma.h | 6 ------ arch/ia64/include/asm/dma.h | 2 -- arch/m68k/include/asm/dma.h | 6 ------ arch/microblaze/include/asm/dma.h | 6 ------ arch/mips/include/asm/dma.h | 8 -------- arch/parisc/include/asm/dma.h | 6 ------ arch/powerpc/include/asm/dma.h | 6 ------ arch/s390/include/asm/dma.h | 6 ------ arch/sh/include/asm/dma.h | 6 ------ arch/sparc/include/asm/dma.h | 8 -------- arch/x86/include/asm/dma.h | 8 -------- arch/xtensa/include/asm/dma.h | 7 ------- drivers/comedi/drivers/comedi_isadma.c | 2 +- drivers/pci/pci.c | 2 ++ drivers/pci/quirks.c | 4 +++- include/linux/isa-dma.h | 14 ++++++++++++++ sound/core/isadma.c | 2 +- 19 files changed, 21 insertions(+), 92 deletions(-) create mode 100644 include/linux/isa-dma.h diff --git a/arch/alpha/include/asm/dma.h b/arch/alpha/include/asm/dma.h index 28610ea7786d..a04d76b96089 100644 --- a/arch/alpha/include/asm/dma.h +++ b/arch/alpha/include/asm/dma.h @@ -365,13 +365,4 @@ extern void free_dma(unsigned int dmanr); /* release it again */ #define KERNEL_HAVE_CHECK_DMA extern int check_dma(unsigned int dmanr); -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - - #endif /* _ASM_DMA_H */ diff --git a/arch/arc/include/asm/dma.h b/arch/arc/include/asm/dma.h index 5b744f4b10a7..02431027ed2f 100644 --- a/arch/arc/include/asm/dma.h +++ b/arch/arc/include/asm/dma.h @@ -7,10 +7,5 @@ #define ASM_ARC_DMA_H #define MAX_DMA_ADDRESS 0xC0000000 -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy 0 -#endif #endif diff --git a/arch/arm/include/asm/dma.h b/arch/arm/include/asm/dma.h index a81dda65c576..907d139be431 100644 --- a/arch/arm/include/asm/dma.h +++ b/arch/arm/include/asm/dma.h @@ -143,10 +143,4 @@ extern int get_dma_residue(unsigned int chan); #endif /* CONFIG_ISA_DMA_API */ -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __ASM_ARM_DMA_H */ diff --git a/arch/ia64/include/asm/dma.h b/arch/ia64/include/asm/dma.h index 59625e9c1f9c..eaed2626ffda 100644 --- a/arch/ia64/include/asm/dma.h +++ b/arch/ia64/include/asm/dma.h @@ -12,8 +12,6 @@ extern unsigned long MAX_DMA_ADDRESS; -extern int isa_dma_bridge_buggy; - #define free_dma(x) #endif /* _ASM_IA64_DMA_H */ diff --git a/arch/m68k/include/asm/dma.h b/arch/m68k/include/asm/dma.h index f6c5e0dfb4e5..1c8d9c5bc2fa 100644 --- a/arch/m68k/include/asm/dma.h +++ b/arch/m68k/include/asm/dma.h @@ -6,10 +6,4 @@ bootmem allocator (but this should do it for this) */ #define MAX_DMA_ADDRESS PAGE_OFFSET -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _M68K_DMA_H */ diff --git a/arch/microblaze/include/asm/dma.h b/arch/microblaze/include/asm/dma.h index f801582be912..7484c9eb66c4 100644 --- a/arch/microblaze/include/asm/dma.h +++ b/arch/microblaze/include/asm/dma.h @@ -9,10 +9,4 @@ /* Virtual address corresponding to last available physical memory address. */ #define MAX_DMA_ADDRESS (CONFIG_KERNEL_START + memory_size - 1) -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_MICROBLAZE_DMA_H */ diff --git a/arch/mips/include/asm/dma.h b/arch/mips/include/asm/dma.h index be726b943530..d6186e6bea7e 100644 --- a/arch/mips/include/asm/dma.h +++ b/arch/mips/include/asm/dma.h @@ -307,12 +307,4 @@ static __inline__ int get_dma_residue(unsigned int dmanr) extern int request_dma(unsigned int dmanr, const char * device_id); /* reserve a DMA channel */ extern void free_dma(unsigned int dmanr); /* release it again */ -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_DMA_H */ diff --git a/arch/parisc/include/asm/dma.h b/arch/parisc/include/asm/dma.h index eea80ed34e6d..9e8c101de902 100644 --- a/arch/parisc/include/asm/dma.h +++ b/arch/parisc/include/asm/dma.h @@ -176,10 +176,4 @@ static __inline__ void set_dma_count(unsigned int dmanr, unsigned int count) #define free_dma(dmanr) -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_DMA_H */ diff --git a/arch/powerpc/include/asm/dma.h b/arch/powerpc/include/asm/dma.h index 6161a9596196..d97c66d9ae34 100644 --- a/arch/powerpc/include/asm/dma.h +++ b/arch/powerpc/include/asm/dma.h @@ -340,11 +340,5 @@ extern int request_dma(unsigned int dmanr, const char *device_id); /* release it again */ extern void free_dma(unsigned int dmanr); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __KERNEL__ */ #endif /* _ASM_POWERPC_DMA_H */ diff --git a/arch/s390/include/asm/dma.h b/arch/s390/include/asm/dma.h index 6f26f35d4a71..dec1c4ce628c 100644 --- a/arch/s390/include/asm/dma.h +++ b/arch/s390/include/asm/dma.h @@ -11,10 +11,4 @@ */ #define MAX_DMA_ADDRESS 0x80000000 -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_S390_DMA_H */ diff --git a/arch/sh/include/asm/dma.h b/arch/sh/include/asm/dma.h index 17d23ae98c77..c8bee3f985a2 100644 --- a/arch/sh/include/asm/dma.h +++ b/arch/sh/include/asm/dma.h @@ -137,10 +137,4 @@ extern int register_chan_caps(const char *dmac, struct dma_chan_caps *capslist); extern int dma_create_sysfs_files(struct dma_channel *, struct dma_info *); extern void dma_remove_sysfs_files(struct dma_channel *, struct dma_info *); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* __ASM_SH_DMA_H */ diff --git a/arch/sparc/include/asm/dma.h b/arch/sparc/include/asm/dma.h index 462e7c794a09..08043f35b110 100644 --- a/arch/sparc/include/asm/dma.h +++ b/arch/sparc/include/asm/dma.h @@ -82,14 +82,6 @@ #define DMA_BURST64 0x40 #define DMA_BURSTBITS 0x7f -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #ifdef CONFIG_SPARC32 struct device; diff --git a/arch/x86/include/asm/dma.h b/arch/x86/include/asm/dma.h index 8e95aa4b0d17..8ae6e0e11b8b 100644 --- a/arch/x86/include/asm/dma.h +++ b/arch/x86/include/asm/dma.h @@ -307,12 +307,4 @@ extern int request_dma(unsigned int dmanr, const char *device_id); extern void free_dma(unsigned int dmanr); #endif -/* From PCI */ - -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - #endif /* _ASM_X86_DMA_H */ diff --git a/arch/xtensa/include/asm/dma.h b/arch/xtensa/include/asm/dma.h index bb099a373b5a..172644539032 100644 --- a/arch/xtensa/include/asm/dma.h +++ b/arch/xtensa/include/asm/dma.h @@ -52,11 +52,4 @@ extern int request_dma(unsigned int dmanr, const char * device_id); extern void free_dma(unsigned int dmanr); -#ifdef CONFIG_PCI -extern int isa_dma_bridge_buggy; -#else -#define isa_dma_bridge_buggy (0) -#endif - - #endif diff --git a/drivers/comedi/drivers/comedi_isadma.c b/drivers/comedi/drivers/comedi_isadma.c index 700982464c53..020b3d1e1ac0 100644 --- a/drivers/comedi/drivers/comedi_isadma.c +++ b/drivers/comedi/drivers/comedi_isadma.c @@ -8,7 +8,7 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/dma-mapping.h> -#include <asm/dma.h> +#include <linux/isa-dma.h> #include <linux/comedi/comedidev.h> #include <linux/comedi/comedi_isadma.h> diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index cfaf40a540a8..60c55d2cb2cc 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -41,8 +41,10 @@ const char *pci_power_names[] = { }; EXPORT_SYMBOL_GPL(pci_power_names); +#ifdef CONFIG_X86_32 int isa_dma_bridge_buggy; EXPORT_SYMBOL(isa_dma_bridge_buggy); +#endif int pci_pci_problems; EXPORT_SYMBOL(pci_pci_problems); diff --git a/drivers/pci/quirks.c b/drivers/pci/quirks.c index 41aeaa235132..6fc64509eee7 100644 --- a/drivers/pci/quirks.c +++ b/drivers/pci/quirks.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/export.h> #include <linux/pci.h> +#include <linux/isa-dma.h> /* isa_dma_bridge_buggy */ #include <linux/init.h> #include <linux/delay.h> #include <linux/acpi.h> @@ -30,7 +31,6 @@ #include <linux/pm_runtime.h> #include <linux/suspend.h> #include <linux/switchtec.h> -#include <asm/dma.h> /* isa_dma_bridge_buggy */ #include "pci.h" static ktime_t fixup_debug_start(struct pci_dev *dev, @@ -239,6 +239,7 @@ static void quirk_passive_release(struct pci_dev *dev) DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82441, quirk_passive_release); DECLARE_PCI_FIXUP_RESUME(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82441, quirk_passive_release); +#ifdef CONFIG_X86_32 /* * The VIA VP2/VP3/MVP3 seem to have some 'features'. There may be a * workaround but VIA don't answer queries. If you happen to have good @@ -265,6 +266,7 @@ DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M1533, quirk_isa_dma DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_1, quirk_isa_dma_hangs); DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_2, quirk_isa_dma_hangs); DECLARE_PCI_FIXUP_FINAL(PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_CBUS_3, quirk_isa_dma_hangs); +#endif /* * Intel NM10 "TigerPoint" LPC PM1a_STS.BM_STS must be clear diff --git a/include/linux/isa-dma.h b/include/linux/isa-dma.h new file mode 100644 index 000000000000..61504a8c1b9e --- /dev/null +++ b/include/linux/isa-dma.h @@ -0,0 +1,14 @@ +/* SPDX-License-Identifier: GPL-2.0 */ + +#ifndef __LINUX_ISA_DMA_H +#define __LINUX_ISA_DMA_H + +#include <asm/dma.h> + +#if defined(CONFIG_PCI) && defined(CONFIG_X86_32) +extern int isa_dma_bridge_buggy; +#else +#define isa_dma_bridge_buggy (0) +#endif + +#endif /* __LINUX_ISA_DMA_H */ diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 1f45ede023b4..18a86212e3a8 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -12,8 +12,8 @@ #undef HAVE_REALLY_SLOW_DMA_CONTROLLER #include <linux/export.h> +#include <linux/isa-dma.h> #include <sound/core.h> -#include <asm/dma.h> /** * snd_dma_program - program an ISA DMA transfer -- 2.36.1

3 2

[PATCH 0/4] ALSA: hda: Unify codec construction
by Cezary Rojewski 20 Jul '22

20 Jul '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… Cezary Rojewski (4): ASoC: hdac_hda: Disconnect struct hdac_hda_priv and hda_codec ALSA: hda: Make snd_hda_codec_device_init() the only codec constructor ALSA: hda: Always free codec on the device release ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 4 +- sound/hda/ext/hdac_ext_bus.c | 34 ++++++-------- sound/pci/hda/hda_codec.c | 49 +++++++++----------- sound/soc/codecs/hdac_hda.c | 26 ++++------- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 24 ++++------ sound/soc/sof/intel/hda-codec.c | 29 +++++------- 10 files changed, 73 insertions(+), 101 deletions(-) -- 2.25.1

4 9

Re: [PATCH] soundwire:repeated word: 'state'.
by Pierre-Louis Bossart 20 Jul '22

20 Jul '22

On 7/19/22 13:39, Xin Gao wrote: > repeated word: 'state'. > > Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> > --- > drivers/soundwire/stream.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c > index d34150559142..f0312b99e3de 100644 > --- a/drivers/soundwire/stream.c > +++ b/drivers/soundwire/stream.c > @@ -1424,7 +1424,7 @@ int sdw_prepare_stream(struct sdw_stream_runtime *stream) > if (stream->state != SDW_STREAM_CONFIGURED && > stream->state != SDW_STREAM_DEPREPARED && > stream->state != SDW_STREAM_DISABLED) { > - pr_err("%s: %s: inconsistent state state %d\n", > + pr_err("%s: %s: inconsistent state %d\n", > __func__, stream->name, stream->state); > ret = -EINVAL; > goto state_err; There are other occurrences of this repetition, you may want to correct them all in the same patch? stream.c:1444: pr_err("%s: %s: inconsistent state state %d\n", stream.c:1533: pr_err("%s: %s: inconsistent state state %d\n", stream.c:1620: pr_err("%s: %s: inconsistent state state %d\n", stream.c:1701: pr_err("%s: %s: inconsistent state state %d\n",

1 0

[PATCH] ALSA: hda: Skip creating captures in SOF context
by Kai-Heng Feng 20 Jul '22

20 Jul '22

On HP laptops that use SOF driver for DMIC, the micmute LED doesn't light up when mic is muted after commit 9b014266ef8a ("ASoC: SOF: topology: use new sound control LED layer"). The micmute LED itself is still working via sysfs, but it doesn't follow mute anymore. That's because unlike vendors like Dell and Lenovo, HP laptops use HDA codec to control mute LEDs instead of ACPI. So on HP laptops, both SOF and HDA create captures with SNDRV_CTL_ELEM_ACCESS_MIC_LED access, snd_ctl_led_set_state() considers there are two different kcontrols and one of them is not muted. So skip creating captures for HDA when it's called from SOF, the captures are already handled by SOF. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_generic.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index fc114e5224806..2a6c1a77c49ac 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5075,10 +5075,14 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, return err; } - - err = create_capture_mixers(codec); - if (err < 0) - return err; + /* Skip creating capture on SOF which creates captures based on + * topology file. + */ + if (codec->core.type != HDA_DEV_ASOC) { + err = create_capture_mixers(codec); + if (err < 0) + return err; + } err = parse_mic_boost(codec); if (err < 0) -- 2.36.1

4 5

[PATCH -next 0/5] ASoC: fsl: Fix sparse warning
by Shengjiu Wang 20 Jul '22

20 Jul '22

Fix sparse warning Shengjiu Wang (5): ASoC: fsl_sai: Don't use plain integer as NULL pointer ASoC: fsl_asrc: force cast the asrc_format type ASoC: fsl-asoc-card: force cast the asrc_format type ASoC: fsl_easrc: use snd_pcm_format_t type for sample_format ASoC: imx-card: use snd_pcm_format_t type for asrc_format sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 ++-- sound/soc/fsl/fsl_easrc.c | 7 ++++--- sound/soc/fsl/fsl_easrc.h | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/imx-card.c | 7 ++++--- 6 files changed, 13 insertions(+), 11 deletions(-) -- 2.34.1

4 12

[PATCH v6] ASoc:dt-bindings:tas2780:remove unused item in DT and revise the format
by Raphael-Xu 20 Jul '22

20 Jul '22

1.remove unused item in DT 2.revise the format Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../devicetree/bindings/sound/tas2780.yaml | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2780.yaml b/Documentation/devicetree/bindings/sound/tas2780.yaml index 5a0de0995066..21389a16fdb9 100644 --- a/Documentation/devicetree/bindings/sound/tas2780.yaml +++ b/Documentation/devicetree/bindings/sound/tas2780.yaml @@ -22,24 +22,29 @@ properties: - ti,tas2780 reg: - maxItems: 1 - description: | - I2C address of the device can be between 0x38 to 0x45. + description: + I2C address of the device can be between 0x38 to 0x3f. + minimum: 0x38 + maximum: 0x3f reset-gpios: - maxItems: 1 description: GPIO used to reset the device. + maxItems: 1 interrupts: maxItems: 1 ti,imon-slot-no: - $ref: /schemas/types.yaml#/definitions/uint32 description: TDM TX current sense time slot. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 ti,vmon-slot-no: - $ref: /schemas/types.yaml#/definitions/uint32 description: TDM TX voltage sense time slot. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 3 '#sound-dai-cells': const: 1 @@ -60,10 +65,8 @@ examples: compatible = "ti,tas2780"; reg = <0x38>; #sound-dai-cells = <1>; - interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; - shutdown-gpios = <&gpio1 15 0>; ti,imon-slot-no = <0>; ti,vmon-slot-no = <2>; }; -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 6/8] ASoC: wm8998: Fix event generation for input mux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 15b2e5d10ccf32a1a1ae7c636511e2f51320fdb5 ] wm8998_inmux_put returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8998.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 17dc5780ab68..3b278435b0aa 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -111,6 +111,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int mode_reg, mode_index; unsigned int mux, inmode, src_val, mode_val; + int change, ret; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) @@ -140,14 +141,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + change = snd_soc_component_update_bits(component, e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); + if (ret < 0) { + dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static const char * const wm8998_inmux_texts[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 12/16] ASoC: cs47l92: Fix event generation for OUT1 demux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 870d72ab9228575b2f005c9a23ea08787e0f63e6 ] cs47l92_put_demux returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs47l92.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index d50f75f3b3e4..cb7d02c3f11b 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum, -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 11/16] ASoC: wm8998: Fix event generation for input mux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 15b2e5d10ccf32a1a1ae7c636511e2f51320fdb5 ] wm8998_inmux_put returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8998.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 817ccddd6344..55b08eae9604 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int mode_reg, mode_index; unsigned int mux, inmode, src_val, mode_val; + int change, ret; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) @@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + change = snd_soc_component_update_bits(component, e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); + if (ret < 0) { + dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static const char * const wm8998_inmux_texts[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 10/16] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Sasha Levin 20 Jul '22

20 Jul '22

From: David Owens <daowens01(a)gmail.com> [ Upstream commit f0d96937d31c4615a6418e4bed5cee50a952040e ] Convert to managed versions of sysfs and clk allocation to simplify unbinding and error handling in probe. Managed sysfs node creation specifically addresses the following error seen the second time probe is attempted after sdma_pcm_platform_register() previously requsted probe deferral: sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' Signed-off-by: David Owens <dowens(a)precisionplanting.com> Link: https://lore.kernel.org/r/20220620183744.3176557-1-dowens@precisionplanting… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp-priv.h | 2 -- sound/soc/ti/omap-mcbsp-st.c | 14 ++------------ sound/soc/ti/omap-mcbsp.c | 19 ++----------------- 3 files changed, 4 insertions(+), 31 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 1a3fe854e856..23dc35192e39 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -350,7 +350,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -362,7 +362,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -371,16 +371,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 3273b317fa3b..8f9d4a2c1d71 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -706,8 +706,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -715,16 +714,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1435,11 +1425,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (pm_qos_request_active(&mcbsp->pm_qos_req)) pm_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 19/25] ASoC: arizona: Update arizona_aif_cfg_changed to use RX_BCLK_RATE
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit f99e930655f411453170a5f332e12c2d2748822e ] Currently the function arizona_aif_cfg_changed uses the TX_BCLK_RATE, however this register is not used on wm8998. This was not noticed as previously snd_soc_component_read did not print an error message. However, now the log gets filled with error messages, further more the test for if the LRCLK changed will return spurious results. Update the code to use the RX_BCLK_RATE register, the LRCLK parameters are written to both registers and the RX_BCLK_RATE register is used across all Arizona devices. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 1228f2de0297..65b0f6585c80 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1759,8 +1759,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component, if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) return true; - val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE); - if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK)) return true; val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 18/25] ASoC: cs47l92: Fix event generation for OUT1 demux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 870d72ab9228575b2f005c9a23ea08787e0f63e6 ] cs47l92_put_demux returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs47l92.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 6e34106c268f..9c22cd197383 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum, -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 17/25] ASoC: wm8998: Fix event generation for input mux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 15b2e5d10ccf32a1a1ae7c636511e2f51320fdb5 ] wm8998_inmux_put returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8998.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 5413254295b7..2491212579f1 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int mode_reg, mode_index; unsigned int mux, inmode, src_val, mode_val; + int change, ret; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) @@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + change = snd_soc_component_update_bits(component, e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); + if (ret < 0) { + dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static const char * const wm8998_inmux_texts[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 16/25] ASoC: tlv320adcx140: Fix tx_mask check
by Sasha Levin 20 Jul '22

20 Jul '22

From: Sascha Hauer <s.hauer(a)pengutronix.de> [ Upstream commit 7d90c8e6396ba245da16bedd789df6d669375408 ] The tx_mask check doesn't reflect what the driver and the chip support. The check currently checks for exactly two slots being enabled. The tlv320adcx140 supports anything between one and eight channels, so relax the check accordingly. The tlv320adcx140 supports arbitrary tx_mask settings, but the driver currently only supports adjacent slots beginning with the first slot, so extend the check to check that the first slot is being used and that there are no holes in the tx_mask. Leave a comment to make it's the driver that limits the tx_mask settings, not the chip itself. While at it remove the set-but-unused struct adcx140p_priv::tdm_delay field. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Link: https://lore.kernel.org/r/20220624105716.2579539-1-s.hauer@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320adcx140.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 53a80246aee1..5579a9053364 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -33,7 +33,6 @@ struct adcx140_priv { bool micbias_vg; unsigned int dai_fmt; - unsigned int tdm_delay; unsigned int slot_width; }; @@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - unsigned int lsb; - /* TDM based on DSP mode requires slots to be adjacent */ - lsb = __ffs(tx_mask); - if ((lsb + 1) != __fls(tx_mask)) { - dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + /* + * The chip itself supports arbitrary masks, but the driver currently + * only supports adjacent slots beginning at the first slot. + */ + if (tx_mask != GENMASK(__fls(tx_mask), 0)) { + dev_err(component->dev, "Only lower adjacent slots are supported\n"); return -EINVAL; } @@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - adcx140->tdm_delay = lsb; adcx140->slot_width = slot_width; return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 15/25] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Sasha Levin 20 Jul '22

20 Jul '22

From: David Owens <daowens01(a)gmail.com> [ Upstream commit f0d96937d31c4615a6418e4bed5cee50a952040e ] Convert to managed versions of sysfs and clk allocation to simplify unbinding and error handling in probe. Managed sysfs node creation specifically addresses the following error seen the second time probe is attempted after sdma_pcm_platform_register() previously requsted probe deferral: sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' Signed-off-by: David Owens <dowens(a)precisionplanting.com> Link: https://lore.kernel.org/r/20220620183744.3176557-1-dowens@precisionplanting… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp-priv.h | 2 -- sound/soc/ti/omap-mcbsp-st.c | 14 ++------------ sound/soc/ti/omap-mcbsp.c | 19 ++----------------- 3 files changed, 4 insertions(+), 31 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..7e8179cae92e 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 6025b30bbe77..119e9053d83f 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -703,8 +703,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -712,16 +711,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1432,11 +1422,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 33/42] ASoC: arizona: Update arizona_aif_cfg_changed to use RX_BCLK_RATE
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit f99e930655f411453170a5f332e12c2d2748822e ] Currently the function arizona_aif_cfg_changed uses the TX_BCLK_RATE, however this register is not used on wm8998. This was not noticed as previously snd_soc_component_read did not print an error message. However, now the log gets filled with error messages, further more the test for if the LRCLK changed will return spurious results. Update the code to use the RX_BCLK_RATE register, the LRCLK parameters are written to both registers and the RX_BCLK_RATE register is used across all Arizona devices. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e32871b3f68a..7434aeeda292 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component, if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) return true; - val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE); - if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK)) return true; val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 32/42] ASoC: cs47l92: Fix event generation for OUT1 demux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 870d72ab9228575b2f005c9a23ea08787e0f63e6 ] cs47l92_put_demux returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs47l92.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 1a0280416d92..ad6f41741372 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum, -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 31/42] ASoC: wm8998: Fix event generation for input mux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 15b2e5d10ccf32a1a1ae7c636511e2f51320fdb5 ] wm8998_inmux_put returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8998.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 00b59fc9b1fe..ab5481187c71 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int mode_reg, mode_index; unsigned int mux, inmode, src_val, mode_val; + int change, ret; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) @@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + change = snd_soc_component_update_bits(component, e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); + if (ret < 0) { + dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static const char * const wm8998_inmux_texts[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 30/42] ASoC: wm5102: Fix event generation for output compensation
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 71b5ab96ffe6589abe7a2e302b83f7a426ebe099 ] The output compensation controls always returns zero regardless of if the control value was updated. This results in missing notifications to user-space of the control change. Update the handling to return 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-1-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 621598608bf0..fc3d4e1a9c55 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + int ret = 0; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + if (arizona->dac_comp_coeff != dac_comp_coeff) { + arizona->dac_comp_coeff = dac_comp_coeff; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, @@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret = 0; + + if (ucontrol->value.integer.value[0] > mc->max) + return -EINVAL; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) { + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static const char * const wm5102_osr_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 29/42] ASoC: tlv320adcx140: Fix tx_mask check
by Sasha Levin 20 Jul '22

20 Jul '22

From: Sascha Hauer <s.hauer(a)pengutronix.de> [ Upstream commit 7d90c8e6396ba245da16bedd789df6d669375408 ] The tx_mask check doesn't reflect what the driver and the chip support. The check currently checks for exactly two slots being enabled. The tlv320adcx140 supports anything between one and eight channels, so relax the check accordingly. The tlv320adcx140 supports arbitrary tx_mask settings, but the driver currently only supports adjacent slots beginning with the first slot, so extend the check to check that the first slot is being used and that there are no holes in the tx_mask. Leave a comment to make it's the driver that limits the tx_mask settings, not the chip itself. While at it remove the set-but-unused struct adcx140p_priv::tdm_delay field. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Link: https://lore.kernel.org/r/20220624105716.2579539-1-s.hauer@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320adcx140.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 32b120d624b2..bf18fcfcaeb9 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -33,7 +33,6 @@ struct adcx140_priv { bool micbias_vg; unsigned int dai_fmt; - unsigned int tdm_delay; unsigned int slot_width; }; @@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - unsigned int lsb; - /* TDM based on DSP mode requires slots to be adjacent */ - lsb = __ffs(tx_mask); - if ((lsb + 1) != __fls(tx_mask)) { - dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + /* + * The chip itself supports arbitrary masks, but the driver currently + * only supports adjacent slots beginning at the first slot. + */ + if (tx_mask != GENMASK(__fls(tx_mask), 0)) { + dev_err(component->dev, "Only lower adjacent slots are supported\n"); return -EINVAL; } @@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - adcx140->tdm_delay = lsb; adcx140->slot_width = slot_width; return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 28/42] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Sasha Levin 20 Jul '22

20 Jul '22

From: David Owens <daowens01(a)gmail.com> [ Upstream commit f0d96937d31c4615a6418e4bed5cee50a952040e ] Convert to managed versions of sysfs and clk allocation to simplify unbinding and error handling in probe. Managed sysfs node creation specifically addresses the following error seen the second time probe is attempted after sdma_pcm_platform_register() previously requsted probe deferral: sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' Signed-off-by: David Owens <dowens(a)precisionplanting.com> Link: https://lore.kernel.org/r/20220620183744.3176557-1-dowens@precisionplanting… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp-priv.h | 2 -- sound/soc/ti/omap-mcbsp-st.c | 14 ++------------ sound/soc/ti/omap-mcbsp.c | 19 ++----------------- 3 files changed, 4 insertions(+), 31 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..7e8179cae92e 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 4479d74f0a45..9933b33c80ca 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 44/54] ASoC: rt5640: Fix the wrong state of JD1 and JD2
by Sasha Levin 20 Jul '22

20 Jul '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 051dade346957d5b68ad986f497835805fa7a9dd ] The patch fixes the wrong state of JD1 and JD2 while the bst1 or bst2 is power on in the HDA JD using. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20220705101134.16792-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 30 ++++++++++++++++++++++-------- 1 file changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 3559d9ecfa07..188c8f401a47 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000); snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000); snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); - snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) + snd_soc_component_write(component, RT5640_PWR_ANLG1, + 0x0018); + else + snd_soc_component_write(component, RT5640_PWR_ANLG1, + 0x0000); snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000); break; @@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work) static irqreturn_t rt5640_irq(int irq, void *data) { struct rt5640_priv *rt5640 = data; + int delay = 0; + + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { + cancel_delayed_work_sync(&rt5640->jack_work); + delay = 100; + } if (rt5640->jack) - queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); + queue_delayed_work(system_long_wq, &rt5640->jack_work, delay); return IRQ_HANDLED; } @@ -2580,6 +2591,12 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); + snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, + RT5640_PWR_VREF2, RT5640_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, + RT5640_PWR_FV2, RT5640_PWR_FV2); + rt5640->jack = jack; ret = request_irq(rt5640->irq, rt5640_irq, @@ -2696,16 +2713,13 @@ static int rt5640_probe(struct snd_soc_component *component) if (device_property_read_u32(component->dev, "realtek,jack-detect-source", &val) == 0) { - if (val <= RT5640_JD_SRC_GPIO4) { + if (val <= RT5640_JD_SRC_GPIO4) rt5640->jd_src = val << RT5640_JD_SFT; - } else if (val == RT5640_JD_SRC_HDA_HEADER) { + else if (val == RT5640_JD_SRC_HDA_HEADER) rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER; - snd_soc_component_update_bits(component, RT5640_DUMMY1, - 0x0300, 0x0); - } else { + else dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n", val); - } } if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted")) -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 43/54] ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph()
by Sasha Levin 20 Jul '22

20 Jul '22

From: Jianglei Nie <niejianglei2021(a)163.com> [ Upstream commit df5b4aca7248dc5a5cae93f162eae0decf972e48 ] q6apm_get_audioreach_graph() allocates a memory chunk for graph->graph with audioreach_alloc_graph_pkt(). When idr_alloc() fails, graph->graph is not released, which will lead to a memory leak. We can release the graph->graph with kfree() when idr_alloc() fails to fix the memory leak. Signed-off-by: Jianglei Nie <niejianglei2021(a)163.com> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20220629182520.2164409-1-niejianglei2021@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/qdsp6/q6apm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index f424d7aa389a..794019286c70 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL); if (id < 0) { dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id); + kfree(graph->graph); kfree(graph); mutex_unlock(&apm->lock); return ERR_PTR(id); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 42/54] ASoC: arizona: Update arizona_aif_cfg_changed to use RX_BCLK_RATE
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit f99e930655f411453170a5f332e12c2d2748822e ] Currently the function arizona_aif_cfg_changed uses the TX_BCLK_RATE, however this register is not used on wm8998. This was not noticed as previously snd_soc_component_read did not print an error message. However, now the log gets filled with error messages, further more the test for if the LRCLK changed will return spurious results. Update the code to use the RX_BCLK_RATE register, the LRCLK parameters are written to both registers and the RX_BCLK_RATE register is used across all Arizona devices. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e32871b3f68a..7434aeeda292 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component, if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) return true; - val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE); - if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK)) return true; val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 41/54] ASoC: cs47l92: Fix event generation for OUT1 demux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 870d72ab9228575b2f005c9a23ea08787e0f63e6 ] cs47l92_put_demux returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs47l92.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index a1b8dcdb9f7b..444026b7d54b 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum, -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 40/54] ASoC: wm8998: Fix event generation for input mux
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 15b2e5d10ccf32a1a1ae7c636511e2f51320fdb5 ] wm8998_inmux_put returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8998.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 00b59fc9b1fe..ab5481187c71 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int mode_reg, mode_index; unsigned int mux, inmode, src_val, mode_val; + int change, ret; mux = ucontrol->value.enumerated.item[0]; if (mux > 1) @@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); - snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | - ARIZONA_IN1L_SRC_SE_MASK, - src_val); + change = snd_soc_component_update_bits(component, e->reg, + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, - ucontrol->value.enumerated.item[0], - e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, + ucontrol->value.enumerated.item[0], + e, NULL); + if (ret < 0) { + dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } static const char * const wm8998_inmux_texts[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 39/54] ASoC: wm5102: Fix event generation for output compensation
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 71b5ab96ffe6589abe7a2e302b83f7a426ebe099 ] The output compensation controls always returns zero regardless of if the control value was updated. This results in missing notifications to user-space of the control change. Update the handling to return 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-1-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index da2f8998df87..b034df47a5ef 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + int ret = 0; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + if (arizona->dac_comp_coeff != dac_comp_coeff) { + arizona->dac_comp_coeff = dac_comp_coeff; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, @@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret = 0; + + if (ucontrol->value.integer.value[0] > mc->max) + return -EINVAL; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) { + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static const char * const wm5102_osr_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 38/54] ASoC: wcd9335: Use int array instead of bitmask for TX mixers
by Sasha Levin 20 Jul '22

20 Jul '22

From: Yassine Oudjana <y.oudjana(a)protonmail.com> [ Upstream commit a5d6d28e2ea38dff017cb562dfbe0259d093a851 ] Currently slim_tx_mixer_get reports all TX mixers as enabled when at least one is, due to it reading the entire tx_port_value bitmask without testing the specific bit corresponding to a TX port. Furthermore, using the same bitmask for all capture DAIs makes setting one mixer affect them all. To prevent this, and since the SLIM TX muxes effectively only connect to one of the mixers at a time, turn tx_port_value into an int array storing the DAI index each of the ports is connected to. Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> Link: https://lore.kernel.org/r/20220622061745.35399-1-y.oudjana@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd9335.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 1e60db4056ad..ad887c9c7573 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -342,7 +342,7 @@ struct wcd9335_codec { struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY]; unsigned int rx_port_value[WCD9335_RX_MAX]; - unsigned int tx_port_value; + unsigned int tx_port_value[WCD9335_TX_MAX]; int hph_l_gain; int hph_r_gain; u32 rx_bias_count; @@ -1328,8 +1328,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev); + struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc); + struct soc_mixer_control *mixer = + (struct soc_mixer_control *)kc->private_value; + int dai_id = widget->shift; + int port_id = mixer->shift; - ucontrol->value.integer.value[0] = wcd->tx_port_value; + ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id; return 0; } @@ -1352,12 +1357,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc, case AIF2_CAP: case AIF3_CAP: /* only add to the list if value not set */ - if (enable && !(wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value |= BIT(port_id); + if (enable && wcd->tx_port_value[port_id] != dai_id) { + wcd->tx_port_value[port_id] = dai_id; list_add_tail(&wcd->tx_chs[port_id].list, &wcd->dai[dai_id].slim_ch_list); - } else if (!enable && (wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value &= ~BIT(port_id); + } else if (!enable && wcd->tx_port_value[port_id] == dai_id) { + wcd->tx_port_value[port_id] = -1; list_del_init(&wcd->tx_chs[port_id].list); } break; -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 37/54] ASoC: tlv320adcx140: Fix tx_mask check
by Sasha Levin 20 Jul '22

20 Jul '22

From: Sascha Hauer <s.hauer(a)pengutronix.de> [ Upstream commit 7d90c8e6396ba245da16bedd789df6d669375408 ] The tx_mask check doesn't reflect what the driver and the chip support. The check currently checks for exactly two slots being enabled. The tlv320adcx140 supports anything between one and eight channels, so relax the check accordingly. The tlv320adcx140 supports arbitrary tx_mask settings, but the driver currently only supports adjacent slots beginning with the first slot, so extend the check to check that the first slot is being used and that there are no holes in the tx_mask. Leave a comment to make it's the driver that limits the tx_mask settings, not the chip itself. While at it remove the set-but-unused struct adcx140p_priv::tdm_delay field. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Link: https://lore.kernel.org/r/20220624105716.2579539-1-s.hauer@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320adcx140.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 32b120d624b2..bf18fcfcaeb9 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -33,7 +33,6 @@ struct adcx140_priv { bool micbias_vg; unsigned int dai_fmt; - unsigned int tdm_delay; unsigned int slot_width; }; @@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - unsigned int lsb; - /* TDM based on DSP mode requires slots to be adjacent */ - lsb = __ffs(tx_mask); - if ((lsb + 1) != __fls(tx_mask)) { - dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + /* + * The chip itself supports arbitrary masks, but the driver currently + * only supports adjacent slots beginning at the first slot. + */ + if (tx_mask != GENMASK(__fls(tx_mask), 0)) { + dev_err(component->dev, "Only lower adjacent slots are supported\n"); return -EINVAL; } @@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - adcx140->tdm_delay = lsb; adcx140->slot_width = slot_width; return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 36/54] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Sasha Levin 20 Jul '22

20 Jul '22

From: David Owens <daowens01(a)gmail.com> [ Upstream commit f0d96937d31c4615a6418e4bed5cee50a952040e ] Convert to managed versions of sysfs and clk allocation to simplify unbinding and error handling in probe. Managed sysfs node creation specifically addresses the following error seen the second time probe is attempted after sdma_pcm_platform_register() previously requsted probe deferral: sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' Signed-off-by: David Owens <dowens(a)precisionplanting.com> Link: https://lore.kernel.org/r/20220620183744.3176557-1-dowens@precisionplanting… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp-priv.h | 2 -- sound/soc/ti/omap-mcbsp-st.c | 14 ++------------ sound/soc/ti/omap-mcbsp.c | 19 ++----------------- 3 files changed, 4 insertions(+), 31 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..7e8179cae92e 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 4479d74f0a45..9933b33c80ca 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } -- 2.35.1

1 0

Behringer UMC204HD HiFi "Default" profile crackling audio
by GitHub issues - opened 20 Jul '22

20 Jul '22

alsa-project/alsa-ucm-conf issue #192 was opened from acepukas: Since the recent update to 1.2.7.2, the Behringer UMC204HD "Default" profile has crackling audio through "Line A" and "Line B" ports. Switching to the "Direct" profile allows for clean audio but it's output over 4 speakers (front left and right, rear left and right). Not really sure how to trouble shoot this. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/192 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASoC: Is this bug ? DAPM source vs sink
by Kuninori Morimoto 20 Jul '22

20 Jul '22

Hi ASoC Cc Lars-Peter I have noticed that DAPM source vs sink are strange. ASoC had used source/sink or input/output before, but it was exchanged by this patch a3423b02cf745c1418f1f199646c450d6fc9ca4e ("ASoC: dapm: Consolidate input and output path handling") I think source/sink pair are converted to SND_SOC_DAPM_DIR_IN/OUT by this patch. source <=> IN sink <=> OUT But, its for_each are inverted #define snd_soc_dapm_widget_for_each_sink_path(w, p) \ snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, p) #define snd_soc_dapm_widget_for_each_source_path(w, p) \ snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_OUT, p) I have thought this for_each is for special handling, but, the original was sink is for sink, source is for source after all. #define snd_soc_dapm_widget_for_each_sink_path(w, p) \ list_for_each_entry(p, &w->sinks, list_source) #define snd_soc_dapm_widget_for_each_source_path(w, p) \ list_for_each_entry(p, &w->sources, list_sink) So, I guess current for_each is wrong, but I'm not 100% sure #define snd_soc_dapm_widget_for_each_sink_path(w, p) \ - snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, p) + snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_OUT, p) #define snd_soc_dapm_widget_for_each_source_path(w, p) \ - snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_OUT, p) + snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, p) I'm happy to create patch if I could get your ack. Thank you for your help !! Best regards --- Kuninori Morimoto

3 3

[PATCH 00/11] ASoC: SOF: Intel: updates and cleanups
by Pierre-Louis Bossart 19 Jul '22

19 Jul '22

Set of updates for IPC3, IPC4, MTL support and cleanups for the topology filename override which was broken for HDaudio platforms. It may be too late for 5.20, it's ok if these patches are applied in the next cycle. The topology override depends on the patch "ASoC: SOF: Intel: enable dmic handling with 2 or fewer SoundWire links" shared in a separate patchset. Bard Liao (1): ASoC: SOF: ipc4-topology: move ida allocate/free to widget_setup/free Peter Ujfalusi (2): ASoC: SOF: ipc3-loader: Print out the non matching ext_man magic number ASoC: SOF: Intel: atom: don't keep a temporary string in fixup_tplg_name Pierre-Louis Bossart (4): ASoC: SOF: Intel: hda: don't set fw_filename ASoC: SOF: Intel: hda: only fixup topology name if not set already ASoC: SOF: probes: rename assign/free callbacks as startup/shutdown ASoC: SOF: Intel: hda-stream: test DMA buffer first in hw_params Rander Wang (3): ASoC: SOF: ipc4-pcm: set pcm rate to dai setting ASoC: SOF: add ipc4 SRC module support ASoC: SOF: ipc4-topology: set domain bit based on dp domain type Yong Zhi (1): ASoC: SOF: Intel: MTL: remove redundant snd_sof_dsp_read() call sound/soc/sof/intel/atom.c | 16 +-- sound/soc/sof/intel/hda-probes.c | 16 +-- sound/soc/sof/intel/hda-stream.c | 15 ++- sound/soc/sof/intel/hda.c | 71 +++++++++----- sound/soc/sof/intel/mtl.c | 13 +-- sound/soc/sof/ipc3-loader.c | 6 +- sound/soc/sof/ipc4-pcm.c | 4 + sound/soc/sof/ipc4-topology.c | 156 +++++++++++++++++++++++++----- sound/soc/sof/ipc4-topology.h | 27 +++++- sound/soc/sof/sof-client-probes.c | 8 +- sound/soc/sof/sof-client-probes.h | 8 +- 11 files changed, 250 insertions(+), 90 deletions(-) -- 2.34.1

2 12

[PATCH] ASOC: nau8821: add new widget to control system clock
by Wallace Lin 19 Jul '22

19 Jul '22

From: Wallace Lin <savagecin0(a)gmail.com> Add new widget to control system clock for power saving. Signed-off-by: Wallace Lin <savagecin0(a)gmail.com> Signed-off-by: Wallace Lin <sjlin0(a)nuvoton.com> --- sound/soc/codecs/nau8821.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 2600be250a3c..83a17c48d594 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -36,6 +36,7 @@ static int nau8821_configure_sysclk(struct nau8821 *nau8821, int clk_id, unsigned int freq); +static bool nau8821_is_jack_inserted(struct regmap *regmap); struct nau8821_fll { int mclk_src; @@ -495,7 +496,33 @@ static int nau8821_output_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int system_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + dev_dbg(nau8821->dev, "system clock control : POWER OFF\n"); + /* Set clock source to disable or internal clock before the + * playback or capture end. Codec needs clock for Jack + * detection and button press if jack inserted; otherwise, + * the clock should be closed. + */ + if (nau8821_is_jack_inserted(nau8821->regmap)) { + nau8821_configure_sysclk(nau8821, + NAU8821_CLK_INTERNAL, 0); + } else { + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + } + } + return 0; +} + static const struct snd_soc_dapm_widget nau8821_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("System Clock", SND_SOC_NOPM, 0, 0, + system_clock_control, SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("MICBIAS", NAU8821_R74_MIC_BIAS, NAU8821_MICBIAS_POWERUP_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC Clock", SND_SOC_NOPM, 0, 0, @@ -607,6 +634,9 @@ static const struct snd_soc_dapm_route nau8821_dapm_routes[] = { {"AIFTX", NULL, "ADCL Digital path"}, {"AIFTX", NULL, "ADCR Digital path"}, + {"AIFTX", NULL, "System Clock"}, + {"AIFRX", NULL, "System Clock"}, + {"DDACL", NULL, "AIFRX"}, {"DDACR", NULL, "AIFRX"}, -- 2.25.1

2 1

[PATCH] ASoC: amd: yc: Decrease level of error message
by Mario Limonciello 19 Jul '22

19 Jul '22

On a number of platforms that contain acp3x controller a new ERR level message is showing up: `acp6x pci device not found` This is because ACP3x and ACP6x share same PCI ID but can be identified by PCI revision. As this is expected behavior for a system with ACP3x decrease message to debug. Fixes: b1630fcbfde6c ("ASoC: amd: yc: add new YC platform varaint support") Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/yc/pci-acp6x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/pci-acp6x.c b/sound/soc/amd/yc/pci-acp6x.c index 20f7a99783f20..77c5fa1f7af14 100644 --- a/sound/soc/amd/yc/pci-acp6x.c +++ b/sound/soc/amd/yc/pci-acp6x.c @@ -159,7 +159,7 @@ static int snd_acp6x_probe(struct pci_dev *pci, case 0x6f: break; default: - dev_err(&pci->dev, "acp6x pci device not found\n"); + dev_dbg(&pci->dev, "acp6x pci device not found\n"); return -ENODEV; } if (pci_enable_device(pci)) { -- 2.25.1

2 1

[PATCH v1 1/1] rsnd: ssiu: add missing .quit callback for gen2
by Jiada Wang 19 Jul '22

19 Jul '22

From: Wang Jiada <jiada_wang(a)mentor.com> commit 83b220cf8eb2a ("ASoC: rsnd: implement BUSIF related code in ssiu.c") moved BUSIF related code in ssiu.c which added .quit callback in ssiu.c to disable error irq when ssiu quits. But .quit callback was added only for gen1 and is missing for gen2, this causes error irq is left enabled after ssiu quits on gen2. This commit adds missing .quit callback for gen2 Signed-off-by: Wang Jiada <jiada_wang(a)mentor.com> Cc: Dean Jenkins <Dean_Jenkins(a)mentor.com> Cc: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/ssiu.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..19a8030a660d 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -417,6 +417,7 @@ static struct rsnd_mod_ops rsnd_ssiu_ops_gen2 = { .name = SSIU_NAME, .dma_req = rsnd_ssiu_dma_req, .init = rsnd_ssiu_init_gen2, + .quit = rsnd_ssiu_quit, .start = rsnd_ssiu_start_gen2, .stop = rsnd_ssiu_stop_gen2, .get_status = rsnd_ssiu_get_status, -- 2.25.1

3 2

[PATCH] docs: driver-api: firmware: add driver firmware guidelines.
by Dave Airlie 19 Jul '22

19 Jul '22

From: Dave Airlie <airlied(a)redhat.com> A recent snafu where Intel ignored upstream feedback on a firmware change, led to a late rc6 fix being required. In order to avoid this in the future we should document some expectations around linux-firmware. I was originally going to write this for drm, but it seems quite generic advice. I'm cc'ing this quite widely to reach subsystems which use fw a lot. Signed-off-by: Dave Airlie <airlied(a)redhat.com> --- Documentation/driver-api/firmware/core.rst | 1 + .../firmware/firmware-usage-guidelines.rst | 34 +++++++++++++++++++ 2 files changed, 35 insertions(+) create mode 100644 Documentation/driver-api/firmware/firmware-usage-guidelines.rst diff --git a/Documentation/driver-api/firmware/core.rst b/Documentation/driver-api/firmware/core.rst index 1d1688cbc078..803cd574bbd7 100644 --- a/Documentation/driver-api/firmware/core.rst +++ b/Documentation/driver-api/firmware/core.rst @@ -13,4 +13,5 @@ documents these features. direct-fs-lookup fallback-mechanisms lookup-order + firmware-usage-guidelines diff --git a/Documentation/driver-api/firmware/firmware-usage-guidelines.rst b/Documentation/driver-api/firmware/firmware-usage-guidelines.rst new file mode 100644 index 000000000000..34d2412e78c6 --- /dev/null +++ b/Documentation/driver-api/firmware/firmware-usage-guidelines.rst @@ -0,0 +1,34 @@ +=================== +Firmware Guidelines +=================== + +Drivers that use firmware from linux-firmware should attempt to follow +the rules in this guide. + +* Firmware should be versioned with at least a major/minor version. It + is suggested that the firmware files in linux-firmware be named with + some device specific name, and just the major version. The + major/minor/patch versions should be stored in a header in the + firmware file for the driver to detect any non-ABI fixes/issues. The + firmware files in linux-firmware should be overwritten with the newest + compatible major version. Newer major version firmware should remain + compatible with all kernels that load that major number. + +* Users should *not* have to install newer firmware to use existing + hardware when they install a newer kernel. If the hardware isn't + enabled by default or under development, this can be ignored, until + the first kernel release that enables that hardware. This means no + major version bumps without the kernel retaining backwards + compatibility for the older major versions. Minor version bumps + should not introduce new features that newer kernels depend on + non-optionally. + +* If a security fix needs lockstep firmware and kernel fixes in order to + be successful, then all supported major versions in the linux-firmware + repo should be updated with the security fix, and the kernel patches + should detect if the firmware is new enough to declare if the security + issue is fixed. All communications around security fixes should point + at both the firmware and kernel fixes. If a security fix requires + deprecating old major versions, then this should only be done as a + last option, and be stated clearly in all communications. + -- 2.36.1

6 7

Re: [PATCH -next v2] ASoC: Intel: Fix missing clk_disable_unprepare() on err in platform_clock_control()
by Pierre-Louis Bossart 19 Jul '22

19 Jul '22

On 7/18/22 20:36, Zhang Zekun wrote: > Fix the missing clk_disable_unprepare() before return > from platform_clock_control() in error handling path. > > Fixes: 9a87fc1e0619 ("ASoC: Intel: bytcr_wm5102: Add machine driver for BYT/WM5102") > Signed-off-by: Zhang Zekun <zhangzekun11(a)huawei.com> > --- > sound/soc/intel/boards/bytcr_wm5102.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c > index 45a6805787f5..3e3cdee4b1ce 100644 > --- a/sound/soc/intel/boards/bytcr_wm5102.c > +++ b/sound/soc/intel/boards/bytcr_wm5102.c > @@ -111,6 +111,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, > ret = byt_wm5102_prepare_and_enable_pll1(codec_dai, 48000); > if (ret) { > dev_err(card->dev, "Error setting codec sysclk: %d\n", ret); > + clk_disable_unprepare(priv->mclk); This change looks legit to me, but you want to CC: maintainers (Mark Brown, Takashi Iwai), use the alsa-devel mailing list and CC: the initial contributor Hans de Goede. > return ret; > } > } else {

1 0

speaker-test channel descriptions don't match expectations
by GitHub issues - opened 19 Jul '22

19 Jul '22

alsa-project/alsa-utils issue #159 was opened from pfps: The text output of speaker-test for -c6 appears to indicate that it is using 5+1 channels, with FL, FR, RL, RR, FC, LFE channels (output in a different order). But the wav files that it uses say FL, FR, RL, ??, FC, Rear Center. And this channel order doesn't seem correct, as the channels are not down-mixed to stereo correctly on my system with pipewire and pipewire-media-session. What can be done to get speaker-test and pipewire to agree on the channels? Issue URL : https://github.com/alsa-project/alsa-utils/issues/159 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1 1/1] ALSA: isa: Use INVALID_HWIRQ definition
by Andy Shevchenko 18 Jul '22

18 Jul '22

Use specific definition for invalid IRQ. It makes the code uniform in respect to the constant used for that. No functional change intended. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/isa/sscape.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 0bc0025f7c19..9adaf91045e9 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -140,8 +140,7 @@ struct soundscape { unsigned char midi_vol; }; -#define INVALID_IRQ ((unsigned)-1) - +#define INVALID_IRQ ((unsigned)INVALID_HWIRQ) static inline struct soundscape *get_card_soundscape(struct snd_card *c) { -- 2.35.1

2 2

Re: linux-next: Tree for Jul 15 (sof-client-ipc-flood-test.c)
by Randy Dunlap 18 Jul '22

18 Jul '22

On 7/15/22 05:52, Stephen Rothwell wrote: > Hi all, > > Changes since 20220714: > on x86_64: ld: vmlinux.o: in function `sof_ipc_flood_probe': sof-client-ipc-flood-test.c:(.text+0xb961d4): undefined reference to `sof_client_get_debugfs_root' ld: vmlinux.o: in function `sof_ipc_flood_dfs_open': sof-client-ipc-flood-test.c:(.text+0xb963ed): undefined reference to `sof_client_get_fw_state' ld: vmlinux.o: in function `sof_ipc_flood_dfs_write': sof-client-ipc-flood-test.c:(.text+0xb9689a): undefined reference to `sof_client_ipc_tx_message' Full randconfig file is attached. -- ~Randy

2 1

[BUG] ASoC: nau8821: Found a bug when removing the module
by Zheyu Ma 18 Jul '22

18 Jul '22

Hello, I found a bug in the driver nau8821. When removing the module, I got a warning: [ 64.226442] nau8821 0-0010: remove [ 64.227621] ------------[ cut here ]------------ [ 64.227918] WARNING: CPU: 5 PID: 243 at kernel/irq/devres.c:144 devm_free_irq+0x80/0x8c [ 64.235307] Call trace: [ 64.235501] devm_free_irq+0x80/0x8c [ 64.235746] nau8821_i2c_remove+0x28/0x3c [snd_soc_nau8821] [ 64.236982] i2c_device_remove+0xcc/0xf8 [ 64.237173] device_release_driver_internal+0x208/0x368 [ 64.237406] driver_detach+0xd8/0x12c [ 64.237583] bus_remove_driver+0x90/0xd0 [ 64.237768] driver_unregister+0x3c/0x6c [ 64.237952] i2c_del_driver+0x54/0x74 [ 64.238127] cleanup_module+0x1c/0x1d0 [snd_soc_nau8821] regards, Zheyu Ma

2 1

[PATCH] ASoC: nau8821: Don't unconditionally free interrupt
by Mark Brown 18 Jul '22

18 Jul '22

The remove() operation unconditionally frees the interrupt for the device but we may not actually have an interrupt so there might be nothing to free. Since the interrupt is requested after all other resources we don't need the explicit free anyway, unwinding is guaranteed to be safe, so just delete the remove() function and let devm take care of things. Reported-by: Zheyu Ma <zheyuma97(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8821.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 2600be250a3c..3ce23c20eca9 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -1699,15 +1699,6 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) return ret; } -static int nau8821_i2c_remove(struct i2c_client *i2c_client) -{ - struct nau8821 *nau8821 = i2c_get_clientdata(i2c_client); - - devm_free_irq(nau8821->dev, nau8821->irq, nau8821); - - return 0; -} - static const struct i2c_device_id nau8821_i2c_ids[] = { { "nau8821", 0 }, { } @@ -1737,7 +1728,6 @@ static struct i2c_driver nau8821_driver = { .acpi_match_table = ACPI_PTR(nau8821_acpi_match), }, .probe_new = nau8821_i2c_probe, - .remove = nau8821_i2c_remove, .id_table = nau8821_i2c_ids, }; module_i2c_driver(nau8821_driver); -- 2.30.2

1 0

[PATCH v2] ASoC: mediatek: mt8195: extract SOF common code
by Chunxu Li 18 Jul '22

18 Jul '22

The functions related to SOF can be reused in different machine drivers, such as mt8195 or mt8186, so extract the common code to avoid duplication. Set mtk_soc_card_data which include machine private data and SOF private data as card drvdata, then the difference between machine private can be ignored such as mt8195_mt6359_priv or mt8186_mt6366_priv, at the same time the SOF related code can be reused in different machine drivers. Signed-off-by: Chunxu Li <chunxu.li(a)mediatek.com> --- sound/soc/mediatek/common/Makefile | 2 +- .../soc/mediatek/common/mtk-dsp-sof-common.c | 196 +++++++++++++++ .../soc/mediatek/common/mtk-dsp-sof-common.h | 36 +++ sound/soc/mediatek/common/mtk-soc-card.h | 17 ++ sound/soc/mediatek/mt8195/mt8195-mt6359.c | 233 ++++-------------- 5 files changed, 301 insertions(+), 183 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dsp-sof-common.c create mode 100644 sound/soc/mediatek/common/mtk-dsp-sof-common.h create mode 100644 sound/soc/mediatek/common/mtk-soc-card.h diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile index acbe01e9e928..576deb7f8cce 100644 --- a/sound/soc/mediatek/common/Makefile +++ b/sound/soc/mediatek/common/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o +snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o obj-$(CONFIG_SND_SOC_MTK_BTCVSD) += mtk-btcvsd.o diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c new file mode 100644 index 000000000000..8b1b623207be --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c @@ -0,0 +1,196 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * mtk-dsp-sof-common.c -- MediaTek dsp sof common ctrl + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Chunxu Li <chunxu.li(a)mediatek.com> + */ + +#include "mtk-dsp-sof-common.h" +#include "mtk-soc-card.h" + +/* fixup the BE DAI link to match any values from topology */ +int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_card *card = rtd->card; + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); + struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + int i, j, ret = 0; + + for (i = 0; i < sof_priv->num_streams; i++) { + struct snd_soc_dai *cpu_dai; + struct snd_soc_pcm_runtime *runtime; + struct snd_soc_dai_link *sof_dai_link = NULL; + const struct sof_conn_stream *conn = &sof_priv->conn_streams[i]; + + if (strcmp(rtd->dai_link->name, conn->normal_link)) + continue; + + for_each_card_rtds(card, runtime) { + if (strcmp(runtime->dai_link->name, conn->sof_link)) + continue; + + for_each_rtd_cpu_dais(runtime, j, cpu_dai) { + if (cpu_dai->stream_active[conn->stream_dir] > 0) { + sof_dai_link = runtime->dai_link; + break; + } + } + break; + } + + if (sof_dai_link && sof_dai_link->be_hw_params_fixup) + ret = sof_dai_link->be_hw_params_fixup(runtime, params); + + break; + } + + return ret; +} +EXPORT_SYMBOL_GPL(mtk_sof_dai_link_fixup); + +int mtk_sof_card_probe(struct snd_soc_card *card) +{ + int i; + struct snd_soc_dai_link *dai_link; + + /* Set stream_name to help sof bind widgets */ + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->no_pcm && !dai_link->stream_name && dai_link->name) + dai_link->stream_name = dai_link->name; + } + + return 0; +} +EXPORT_SYMBOL_GPL(mtk_sof_card_probe); + +int mtk_sof_card_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_component *sof_comp = NULL; + struct mtk_soc_card_data *soc_card_data = + snd_soc_card_get_drvdata(card); + struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + int i; + + /* 1. find sof component */ + for_each_card_rtds(card, rtd) { + sof_comp = snd_soc_rtdcom_lookup(rtd, "sof-audio-component"); + if (sof_comp) + break; + } + + if (!sof_comp) { + dev_info(card->dev, "probe without sof-audio-component\n"); + return 0; + } + + /* 2. add route path and fixup callback */ + for (i = 0; i < sof_priv->num_streams; i++) { + const struct sof_conn_stream *conn = &sof_priv->conn_streams[i]; + struct snd_soc_pcm_runtime *sof_rtd = NULL; + struct snd_soc_pcm_runtime *normal_rtd = NULL; + + for_each_card_rtds(card, rtd) { + if (!strcmp(rtd->dai_link->name, conn->sof_link)) { + sof_rtd = rtd; + continue; + } + if (!strcmp(rtd->dai_link->name, conn->normal_link)) { + normal_rtd = rtd; + continue; + } + if (normal_rtd && sof_rtd) + break; + } + if (normal_rtd && sof_rtd) { + int j; + struct snd_soc_dai *cpu_dai; + + for_each_rtd_cpu_dais(sof_rtd, j, cpu_dai) { + struct snd_soc_dapm_route route; + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_widget *play_widget = + cpu_dai->playback_widget; + struct snd_soc_dapm_widget *cap_widget = + cpu_dai->capture_widget; + memset(&route, 0, sizeof(route)); + if (conn->stream_dir == SNDRV_PCM_STREAM_CAPTURE && + cap_widget) { + snd_soc_dapm_widget_for_each_sink_path(cap_widget, p) { + route.source = conn->sof_dma; + route.sink = p->sink->name; + snd_soc_dapm_add_routes(&card->dapm, &route, 1); + } + } else if (conn->stream_dir == SNDRV_PCM_STREAM_PLAYBACK && + play_widget) { + snd_soc_dapm_widget_for_each_source_path(play_widget, p) { + route.source = p->source->name; + route.sink = conn->sof_dma; + snd_soc_dapm_add_routes(&card->dapm, &route, 1); + } + } else { + dev_err(cpu_dai->dev, "stream dir and widget not pair\n"); + } + } + + sof_rtd->dai_link->be_hw_params_fixup = + sof_comp->driver->be_hw_params_fixup; + if (sof_priv->sof_dai_link_fixup) + normal_rtd->dai_link->be_hw_params_fixup = + sof_priv->sof_dai_link_fixup; + else + normal_rtd->dai_link->be_hw_params_fixup = mtk_sof_dai_link_fixup; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(mtk_sof_card_late_probe); + +int mtk_sof_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, + const char *propname, struct snd_soc_dai_link *pre_dai_links, + int pre_num_links) +{ + struct device *dev = card->dev; + struct snd_soc_dai_link *parsed_dai_link; + const char *dai_name = NULL; + int i, j, ret, num_links, parsed_num_links = 0; + + num_links = of_property_count_strings(np, "mediatek,dai-link"); + if (num_links < 0 || num_links > card->num_links) { + dev_dbg(dev, "number of dai-link is invalid\n"); + return -EINVAL; + } + + parsed_dai_link = devm_kcalloc(dev, num_links, sizeof(*parsed_dai_link), GFP_KERNEL); + if (!parsed_dai_link) + return -ENOMEM; + + for (i = 0; i < num_links; i++) { + ret = of_property_read_string_index(np, propname, i, &dai_name); + if (ret) { + dev_dbg(dev, "ASoC: Property '%s' index %d could not be read: %d\n", + propname, i, ret); + return ret; + } + dev_dbg(dev, "ASoC: Property get dai_name:%s\n", dai_name); + for (j = 0; j < pre_num_links; j++) { + if (!strcmp(dai_name, pre_dai_links[j].name)) { + memcpy(&parsed_dai_link[parsed_num_links++], &pre_dai_links[j], + sizeof(struct snd_soc_dai_link)); + break; + } + } + } + + if (parsed_num_links != num_links) + return -EINVAL; + + card->dai_link = parsed_dai_link; + card->num_links = parsed_num_links; + + return 0; +} +EXPORT_SYMBOL_GPL(mtk_sof_dailink_parse_of); diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.h b/sound/soc/mediatek/common/mtk-dsp-sof-common.h new file mode 100644 index 000000000000..dd38c4a93574 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.h @@ -0,0 +1,36 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * mtk-dsp-sof-common.h -- MediaTek dsp sof common definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Chunxu Li <chunxu.li(a)mediatek.com> + */ + +#ifndef _MTK_DSP_SOF_COMMON_H_ +#define _MTK_DSP_SOF_COMMON_H_ + +#include <sound/soc.h> + +struct sof_conn_stream { + const char *normal_link; + const char *sof_link; + const char *sof_dma; + int stream_dir; +}; + +struct mtk_sof_priv { + const struct sof_conn_stream *conn_streams; + int num_streams; + int (*sof_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); +}; + +int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); +int mtk_sof_card_probe(struct snd_soc_card *card); +int mtk_sof_card_late_probe(struct snd_soc_card *card); +int mtk_sof_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, + const char *propname, struct snd_soc_dai_link *pre_dai_links, + int pre_num_links); + +#endif diff --git a/sound/soc/mediatek/common/mtk-soc-card.h b/sound/soc/mediatek/common/mtk-soc-card.h new file mode 100644 index 000000000000..eeda79370049 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-soc-card.h @@ -0,0 +1,17 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * mtk-soc-card.h -- MediaTek soc card data definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Chunxu Li <chunxu.li(a)mediatek.com> + */ + +#ifndef _MTK_SOC_CARD_H_ +#define _MTK_SOC_CARD_H_ + +struct mtk_soc_card_data { + void *mach_priv; + void *sof_priv; +}; + +#endif diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 54a00b0699b1..c530e3fc27e4 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -20,6 +20,8 @@ #include "../../codecs/rt1011.h" #include "../../codecs/rt5682.h" #include "../common/mtk-afe-platform-driver.h" +#include "../common/mtk-dsp-sof-common.h" +#include "../common/mtk-soc-card.h" #include "mt8195-afe-clk.h" #include "mt8195-afe-common.h" @@ -54,13 +56,6 @@ struct mt8195_card_data { unsigned long quirk; }; -struct sof_conn_stream { - const char *normal_link; - const char *sof_link; - const char *sof_dma; - int stream_dir; -}; - struct mt8195_mt6359_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; @@ -374,7 +369,8 @@ static const struct snd_soc_ops mt8195_dptx_ops = { static int mt8195_dptx_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; int ret; @@ -389,7 +385,8 @@ static int mt8195_dptx_codec_init(struct snd_soc_pcm_runtime *rtd) static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; int ret; @@ -555,7 +552,8 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; struct snd_soc_jack *jack = &priv->headset_jack; struct snd_soc_component *cmpnt_afe = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); @@ -722,7 +720,8 @@ static int mt8195_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { struct snd_soc_component *component = dapm->component; - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; int ret; /* @@ -1321,175 +1320,24 @@ static struct snd_soc_card mt8195_mt6359_soc_card = { static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai_link *sof_dai_link = NULL; - struct snd_soc_pcm_runtime *runtime; - struct snd_soc_dai *cpu_dai; - int i, j, ret = 0; - - for (i = 0; i < ARRAY_SIZE(g_sof_conn_streams); i++) { - const struct sof_conn_stream *conn = &g_sof_conn_streams[i]; - - if (strcmp(rtd->dai_link->name, conn->normal_link)) - continue; - - for_each_card_rtds(card, runtime) { - if (strcmp(runtime->dai_link->name, conn->sof_link)) - continue; - - for_each_rtd_cpu_dais(runtime, j, cpu_dai) { - if (cpu_dai->stream_active[conn->stream_dir] > 0) { - sof_dai_link = runtime->dai_link; - break; - } - } - break; - } + int ret; - if (sof_dai_link && sof_dai_link->be_hw_params_fixup) - ret = sof_dai_link->be_hw_params_fixup(runtime, params); - - break; - } + ret = mtk_sof_dai_link_fixup(rtd, params); if (!strcmp(rtd->dai_link->name, "ETDM2_IN_BE") || !strcmp(rtd->dai_link->name, "ETDM1_OUT_BE")) { - mt8195_etdm_hw_params_fixup(runtime, params); + mt8195_etdm_hw_params_fixup(rtd, params); } return ret; } -static int mt8195_mt6359_card_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *runtime; - struct snd_soc_component *sof_comp = NULL; - int i; - - /* 1. find sof component */ - for_each_card_rtds(card, runtime) { - for (i = 0; i < runtime->num_components; i++) { - if (!runtime->components[i]->driver->name) - continue; - if (!strcmp(runtime->components[i]->driver->name, "sof-audio-component")) { - sof_comp = runtime->components[i]; - break; - } - } - } - - if (!sof_comp) { - dev_info(card->dev, " probe without component\n"); - return 0; - } - /* 2. add route path and fixup callback */ - for (i = 0; i < ARRAY_SIZE(g_sof_conn_streams); i++) { - const struct sof_conn_stream *conn = &g_sof_conn_streams[i]; - struct snd_soc_pcm_runtime *sof_rtd = NULL; - struct snd_soc_pcm_runtime *normal_rtd = NULL; - struct snd_soc_pcm_runtime *rtd = NULL; - - for_each_card_rtds(card, rtd) { - if (!strcmp(rtd->dai_link->name, conn->sof_link)) { - sof_rtd = rtd; - continue; - } - if (!strcmp(rtd->dai_link->name, conn->normal_link)) { - normal_rtd = rtd; - continue; - } - if (normal_rtd && sof_rtd) - break; - } - if (normal_rtd && sof_rtd) { - int j; - struct snd_soc_dai *cpu_dai; - - for_each_rtd_cpu_dais(sof_rtd, j, cpu_dai) { - struct snd_soc_dapm_route route; - struct snd_soc_dapm_path *p = NULL; - struct snd_soc_dapm_widget *play_widget = - cpu_dai->playback_widget; - struct snd_soc_dapm_widget *cap_widget = - cpu_dai->capture_widget; - memset(&route, 0, sizeof(route)); - if (conn->stream_dir == SNDRV_PCM_STREAM_CAPTURE && - cap_widget) { - snd_soc_dapm_widget_for_each_sink_path(cap_widget, p) { - route.source = conn->sof_dma; - route.sink = p->sink->name; - snd_soc_dapm_add_routes(&card->dapm, &route, 1); - } - } else if (conn->stream_dir == SNDRV_PCM_STREAM_PLAYBACK && - play_widget){ - snd_soc_dapm_widget_for_each_source_path(play_widget, p) { - route.source = p->source->name; - route.sink = conn->sof_dma; - snd_soc_dapm_add_routes(&card->dapm, &route, 1); - } - } else { - dev_err(cpu_dai->dev, "stream dir and widget not pair\n"); - } - } - normal_rtd->dai_link->be_hw_params_fixup = mt8195_dai_link_fixup; - } - } - - return 0; -} - -static int mt8195_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, - const char *propname) -{ - struct device *dev = card->dev; - struct snd_soc_dai_link *link; - const char *dai_name = NULL; - int i, j, ret, num_links; - - num_links = of_property_count_strings(np, "mediatek,dai-link"); - - if (num_links < 0 || num_links > ARRAY_SIZE(mt8195_mt6359_dai_links)) { - dev_dbg(dev, "number of dai-link is invalid\n"); - return -EINVAL; - } - - card->dai_link = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL); - if (!card->dai_link) - return -ENOMEM; - - card->num_links = 0; - link = card->dai_link; - - for (i = 0; i < num_links; i++) { - ret = of_property_read_string_index(np, propname, i, &dai_name); - if (ret) { - dev_dbg(dev, "ASoC: Property '%s' index %d could not be read: %d\n", - propname, i, ret); - return -EINVAL; - } - - for (j = 0; j < ARRAY_SIZE(mt8195_mt6359_dai_links); j++) { - if (!strcmp(dai_name, mt8195_mt6359_dai_links[j].name)) { - memcpy(link, &mt8195_mt6359_dai_links[j], - sizeof(struct snd_soc_dai_link)); - link++; - card->num_links++; - break; - } - } - } - - if (card->num_links != num_links) - return -EINVAL; - - return 0; -} - static int mt8195_mt6359_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8195_mt6359_soc_card; struct snd_soc_dai_link *dai_link; - struct mt8195_mt6359_priv *priv; + struct mtk_soc_card_data *soc_card_data; + struct mt8195_mt6359_priv *mach_priv; struct device_node *platform_node, *adsp_node, *dp_node, *hdmi_node; struct mt8195_card_data *card_data; int is5682s = 0; @@ -1512,17 +1360,41 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) if (strstr(card->name, "_5682s")) is5682s = 1; + soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL); + if (!soc_card_data) + return -ENOMEM; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) + mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL); + if (!mach_priv) return -ENOMEM; + soc_card_data->mach_priv = mach_priv; + + adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); + if (adsp_node) { + struct mtk_sof_priv *sof_priv; + + sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); + if (!sof_priv) { + ret = -ENOMEM; + goto err_kzalloc; + } + sof_priv->conn_streams = g_sof_conn_streams; + sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); + sof_priv->sof_dai_link_fixup = mt8195_dai_link_fixup; + soc_card_data->sof_priv = sof_priv; + card->late_probe = mtk_sof_card_late_probe; + sof_on = 1; + } + if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { - ret = mt8195_dailink_parse_of(card, pdev->dev.of_node, - "mediatek,dai-link"); + ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, + "mediatek,dai-link", + mt8195_mt6359_dai_links, + ARRAY_SIZE(mt8195_mt6359_dai_links)); if (ret) { dev_dbg(&pdev->dev, "Parse dai-link fail\n"); - return -EINVAL; + goto err_parse_of; } } else { if (!sof_on) @@ -1533,13 +1405,10 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) "mediatek,platform", 0); if (!platform_node) { dev_dbg(&pdev->dev, "Property 'platform' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto err_platform_node; } - adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); - if (adsp_node) - sof_on = 1; - dp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,dptx-codec", 0); hdmi_node = of_parse_phandle(pdev->dev.of_node, "mediatek,hdmi-codec", 0); @@ -1612,17 +1481,17 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) } } - if (sof_on) - card->late_probe = mt8195_mt6359_card_late_probe; - - snd_soc_card_set_drvdata(card, priv); + snd_soc_card_set_drvdata(card, soc_card_data); ret = devm_snd_soc_register_card(&pdev->dev, card); of_node_put(platform_node); - of_node_put(adsp_node); of_node_put(dp_node); of_node_put(hdmi_node); +err_kzalloc: +err_parse_of: +err_platform_node: + of_node_put(adsp_node); return ret; } -- 2.25.1

2 1

FAIL: alsa/Makefile dependency check
by li_jessen2016＠gmail.com li 18 Jul '22

18 Jul '22

Dear Mark Brown, Sorry to bother you. I posted this issue on stackoverflow but didn't get any correct answer yet. So I try to reach you. I'm using Ubuntu 20.04 and the kernel is *5.19.0-rc3+*. When I run *./kselftest_deps.sh gcc* in .*/tools/testing/selftests*, it shows the following: FAIL: alsa/Makefile dependency check: $(shell FAIL: alsa/Makefile dependency check: pkg-config FAIL: alsa/Makefile dependency check: --libs FAIL: alsa/Makefile dependency check: alsa) FAIL: alsa/Makefile dependency check: $(shell FAIL: alsa/Makefile dependency check: pkg-config FAIL: alsa/Makefile dependency check: --libs FAIL: alsa/Makefile dependency check: alsa) In alsa/Makefile, it is: LDLIBS += $(shell pkg-config --libs alsa) ifeq ($(LDLIBS),) LDLIBS += -lasound endif But in ./alsa directory, when I run *$pkg-config --libs alsa, *it returns *-lasound**.* And when I change *$(shell pkg-config --libs alsa)* to *-lasound* in alsa/Makefile, there is no FAIL info for it. So I wonder why the FAIL info appears in the presence of *$(shell pkg-config --libs alsa) *in alsa/Makefile. Is it some sort of bug or did I miss something? Looking forward to your answer. Best Regards. Jessen

2 3

ucm2 profile for MOTU M2
by GitHub pull_request - opened 18 Jul '22

18 Jul '22

alsa-project/alsa-ucm-conf pull request #191 was opened from tw1618: With Release v1.2.7.2 my Audio Interface Motu M2 stopped working as discussed in #190. It looks like that the Motu M4 config #158 is loaded as they use the same (?) ID Those changes work for me. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9128719/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/191 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/191.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH AUTOSEL 5.15 01/28] ASoC: ops: Fix off by one in range control validation
by Sasha Levin 18 Jul '22

18 Jul '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 5871321fb4558c55bf9567052b618ff0be6b975e ] We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index f32ba64c5dda..e73360e9de8f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -526,7 +526,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -547,7 +547,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.35.1

2 21

USB-Audio: Motu M2: Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - edited 17 Jul '22

17 Jul '22

alsa-project/alsa-ucm-conf issue #190 was edited from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/190 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - edited 17 Jul '22

17 Jul '22

alsa-project/alsa-ucm-conf issue #190 was edited from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/190 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-ucm-conf issue #190 was opened from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/190 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH alsa-utils] aplay/aplay.c: make UCM support optional
by Thomas Petazzoni 17 Jul '22

17 Jul '22

Commit 90f59671784a7e47b40485095cd66892d4840ed7 ("topology: pre-processor: Move the call to expand variables") modified aplay by unconditionally invoking features of the use case manager (UCM) from alsa-lib. However, alsa-lib can be compiled with UCM support. In order to properly support this situation, this commit changes aplay to only conditionally compile the UCM related code. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- aplay/aplay.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/aplay/aplay.c b/aplay/aplay.c index b3b3635..59f937d 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -41,7 +41,9 @@ #include <time.h> #include <locale.h> #include <alsa/asoundlib.h> +#ifdef HAVE_ALSA_USE_CASE_H #include <alsa/use-case.h> +#endif #include <assert.h> #include <termios.h> #include <signal.h> @@ -453,6 +455,7 @@ static ssize_t xwrite(int fd, const void *buf, size_t count) return offset; } +#ifdef HAVE_ALSA_USE_CASE_H static int open_ucm(snd_use_case_mgr_t **uc_mgr, char **pcm_name, const char *name) { char *s, *p; @@ -476,6 +479,7 @@ static int open_ucm(snd_use_case_mgr_t **uc_mgr, char **pcm_name, const char *na } return err; } +#endif static long parse_long(const char *str, int *err) { @@ -553,7 +557,9 @@ int main(int argc, char *argv[]) int do_device_list = 0, do_pcm_list = 0, force_sample_format = 0; snd_pcm_info_t *info; FILE *direction; +#ifdef HAVE_ALSA_USE_CASE_H snd_use_case_mgr_t *uc_mgr = NULL; +#endif #ifdef ENABLE_NLS setlocale(LC_ALL, ""); @@ -852,6 +858,7 @@ int main(int argc, char *argv[]) goto __end; } +#ifdef HAVE_ALSA_USE_CASE_H if (strncmp(pcm_name, "ucm.", 4) == 0) { err = open_ucm(&uc_mgr, &pcm_name, pcm_name + 4); if (err < 0) { @@ -861,6 +868,7 @@ int main(int argc, char *argv[]) if (verbose) fprintf(stderr, _("Found UCM PCM device: %s\n"), pcm_name); } +#endif err = snd_pcm_open(&handle, pcm_name, stream, open_mode); if (err < 0) { @@ -951,8 +959,10 @@ int main(int argc, char *argv[]) if (verbose==2) putchar('\n'); snd_pcm_close(handle); +#ifdef HAVE_ALSA_USE_CASE_H if (uc_mgr) snd_use_case_mgr_close(uc_mgr); +#endif handle = NULL; free(audiobuf); __end: -- 2.36.1

1 0

[PATCH alsa-utils] alsactl/info.c: fix conditionals on __ALSA_PCM_H and __ALSA_RAWMIDI_H
by Thomas Petazzoni 17 Jul '22

17 Jul '22

Commit bbc74a61ac7c35e506c3d7f76ecf943cb55736a6 ("alsactl: implement 'info' command") implemented an alsactl info command. In this implementation, there was an attempt to properly address optional features from alsa-lib by using conditions on __ALSA_PCM_H, __ALSA_RAWMIDI_H. Unfortunately, this attempt does not work entirely: only the code inside pcm_device_list(), rawmidi_device_list() was conditionally compiled, but their very prototype also use type definitions provided in pcm.h and rawmidi.h. So really, it's the entire function that needs to be conditionally implemented. Also, snd_rawmidi_stream_name() was not handled properly, for the same reason. This commit implements pcm_device_list() only if __ALSA_PCM_H is defined, and implements snd_rawmidi_stream_name() and rawmidi_device_list() only if __ALSA_RAWMIDI_H is defined. general_card_info() is modified to not call the PCM or raw MIDI functions when support is not available. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- alsactl/info.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/alsactl/info.c b/alsactl/info.c index 253539d..9bd72af 100644 --- a/alsactl/info.c +++ b/alsactl/info.c @@ -22,9 +22,9 @@ #include "aconfig.h" #include "alsactl.h" +#ifdef __ALSA_PCM_H static int pcm_device_list(snd_ctl_t *ctl, snd_pcm_stream_t stream, bool *first) { -#ifdef __ALSA_PCM_H int err, dev, idx; unsigned int count; snd_pcm_info_t *pcminfo; @@ -76,10 +76,12 @@ static int pcm_device_list(snd_ctl_t *ctl, snd_pcm_stream_t stream, bool *first) idx, snd_pcm_info_get_subdevice_name(pcminfo)); } } -#endif + return 0; } +#endif +#ifdef __ALSA_RAWMIDI_H static const char *snd_rawmidi_stream_name(snd_rawmidi_stream_t stream) { if (stream == SND_RAWMIDI_STREAM_INPUT) @@ -91,7 +93,6 @@ static const char *snd_rawmidi_stream_name(snd_rawmidi_stream_t stream) static int rawmidi_device_list(snd_ctl_t *ctl, snd_rawmidi_stream_t stream, bool *first) { -#ifdef __ALSA_RAWMIDI_H int err, dev, idx; unsigned int count; snd_rawmidi_info_t *info; @@ -143,9 +144,10 @@ static int rawmidi_device_list(snd_ctl_t *ctl, snd_rawmidi_stream_t stream, bool idx, snd_rawmidi_info_get_subdevice_name(info)); } } -#endif + return 0; } +#endif static int hwdep_device_list(snd_ctl_t *ctl) { @@ -227,17 +229,21 @@ int general_card_info(int cardno) } err = card_info(ctl); +#ifdef __ALSA_PCM_H first = true; if (err >= 0) err = pcm_device_list(ctl, SND_PCM_STREAM_PLAYBACK, &first); if (err >= 0) err = pcm_device_list(ctl, SND_PCM_STREAM_CAPTURE, &first); +#endif +#ifdef __ALSA_RAWMIDI_H first = true; if (err >= 0) err = rawmidi_device_list(ctl, SND_PCM_STREAM_PLAYBACK, &first); if (err >= 0) err = rawmidi_device_list(ctl, SND_PCM_STREAM_CAPTURE, &first); +#endif if (err >= 0) err = hwdep_device_list(ctl); -- 2.36.1

1 0

Re: [PATCH 6/6] i2c: Make remove callback return void
by Pavel Machek 17 Jul '22

17 Jul '22

Hi! > From: Uwe Kleine-König <uwe(a)kleine-koenig.org> > > The value returned by an i2c driver's remove function is mostly ignored. > (Only an error message is printed if the value is non-zero that the > error is ignored.) > > So change the prototype of the remove function to return no value. This > way driver authors are not tempted to assume that passing an error to > the upper layer is a good idea. All drivers are adapted accordingly. > There is no intended change of behaviour, all callbacks were prepared to > return 0 before. > > Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> 2-4: Acked-by: Pavel Machek <pavel(a)ucw.cz> Best regards, Pavel -- People of Russia, stop Putin before his war on Ukraine escalates.

1 0

Inappropriate Envy24 rate locking/XRUNS for playback and/or recording
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-tools issue #11 was opened from foobers: On Debian 'oldstable', there's a rate-locking and 'low pitch' problem with following setup (same on Mint/Unstable): - dedicated audio card: Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) based - integrated audio in graphics card: Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device xxxx - onboard audio chip: Audio device: Advanced Micro Devices, Inc. [AMD/ATI] xxx (Intel HDA) (xxxx) Initially, in 'alsamixer', there are two values shown for 'Multi Track Internal Clock': '[44100]' and 'Default [41000]' for the dedicated AC. In alsamixer, it's only possible, to change the first value, but without any noticable effect (only using 'Envy24 Control' or 'Mudita24' seem to have). Now, using the audio card, sometimes, right from the start, playback of internet audio streams (in Firefox) seems to have a wrong sampling rate set (i. e. probably 48 kHz), resulting in audibly lower 'pitch' and sound distortion. Even worse, playback isn't always affected from start, but recordings from 'gnome-sound-recorder' will be in either case. This can result in several hours of recorded audio being borked, since there's no controlling of the outcome, until recording is stopped, which is very undesirable, especially in ad-hoc recording situations. I was able to track down the problem being linked to 'Master clock', 'Rate state' settings 'Locked' and 'Reset' in 'Envy24 Control Utility': checking and unchecking / switching forth to 48kHz and back to 44100Hz, helped resetting both playback and recording sampling rate to 44100Hz, to perform regular audio recordings with 'gnome-sound-recorder'. The rate locking problem can also occur, after playing MP4 videos with Mplayer (media info: sampling rate 44100Hz); then, audio files (media info: bit rate 112Kbit/s, sampling rate 44100Hz), previously kown-good audio records will be played at a noticably lower 'pitch', also recording (with gnome-recorder) now will record at a lower pitch, i. e wrong rate. In alsamixer, there are now two different values shown for 'Multi Track Internal Clock': '[44100]' and 'Default [48000]'. Also, when trying to play a regular film DVD with VLC now, it very likely shows short dropouts of black screen and audio, at steady intervals of ~2s, requiring performing a logoff and login, to reset rates again. Interestingly (and unexpectedly), setting 'Master clock' to 'S/PDIF in', seemed to internally freeze the clock / source to 'S/PDIF', in effect preventing any more sound output or recording from the device; trying to switch back to 44110Hz or 48000Hz would fail and the could only be reset, after until doing a complete reboot. I've seen references, pointing to the rate locking issue, which suggest, that unchecking 'locked' and 'reset' for 'Rate state' in 'Envy24' would be the key here - only, on my system, those values didn't seem to be initially checked (at least not visually): https://discourse.ardour.org/t/all-sound-is-low-pitched/82149 So the solution here would seem to be, to ensure unset 'Rate state' options at boot time; in contrast to this suggestion, the documentation says, 'locked' might cause 'errors' and 'XRUNS', and 'reset' only would be the recommended setting by default, to allow applications alter the clock sample rate to their needs, resetting them to the selected 'Master clock' default afterwards: https://wiki.archlinux.org/title/Envy24control#Rate_State While this seems to make sense, 'reset' not being set appropriatly, might also explain some of the weird behavior described above, namely, when some application sets and locks 48kHz for playback and/or input channels, which then don't match the input and/or output of another application, started after that. The effect of VLC/MPlayer showing 'dropouts' may also have its cause in the mentioned 'XRUNS', in this case, supposedly buffer underruns, when applying an 48kHz sampling rate to video files with 44,1kHz audio streams, or buffer overflows, when applying 44,1kHz sampling to files with 48kHz audio streams: https://unix.stackexchange.com/questions/199498/what-are-xruns What (combination) of 'Rate state' settings (if any) would be a 'bullet-proof' remedy here, I cannot tell, since my settings obviously both empty. So, either this was a false visual feedback, or 'reset' does not work as intended. So far, I can't confirm, if these settings differ in Mint, where the problem shows, too, as said (or if the defaults have been altered). Currently I can't try perma-setting 'reset' only (the recommended default), but I think it didn't work out before for recording. Issue URL : https://github.com/alsa-project/alsa-tools/issues/11 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH] ASoC: DPCM: Don't pick up BE without substream
by Takashi Iwai 17 Jul '22

17 Jul '22

When DPCM tries to add valid BE connections at dpcm_add_paths(), it doesn't check whether the picked BE actually supports for the given stream direction. Due to that, when an asymmetric BE stream is present, it picks up wrongly and this may result in a NULL dereference at a later point where the code assumes the existence of a corresponding BE substream. This patch adds the check for the presence of the substream for the target BE for avoiding the problem above. Note that we have already some fix for non-existing BE substream at commit 6246f283d5e0 ("ASoC: dpcm: skip missing substream while applying symmetry"). But the code path we've hit recently is rather happening before the previous fix. So this patch tries to fix at picking up a BE instead of parsing BE lists. Fixes: bbf7d3b1c4f4 ("ASoC: soc-pcm: align BE 'atomicity' with that of the FE") Reported-by: Alex Natalsson <harmoniesworlds(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/CADs9LoPZH_D+eJ9qjTxSLE5jGyhKsjMN7g2NighZ16biVxsy… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..0c1de5624842 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1318,6 +1318,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; + if (!snd_soc_dpcm_get_substream(be, stream)) + continue; + for_each_rtd_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); -- 2.35.3

1 0

Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-lib issue #254 was opened from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-lib/issues/254 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/5] ALSA: Drop async signal support
by Takashi Iwai 16 Jul '22

16 Jul '22

Hi, this is a patch series to drop one of unused features, the async signal support, from ALSA core. The async signal itself is very difficult to use properly due to various restrictions (e.g. you cannot perform any I/O in the context), hence it's a feature that has been never used by real applications. OTOH, the real problem is that there have been quite a few syzcaller reports indicating that fasync code path may lead to some potential deadlocks for long time. Dropping the feature is the easiest solution, obviously. The corresponding update for alsa-lib will follow once when we agree with this approach. thanks, Takashi === Takashi Iwai (5): ALSA: timer: Drop async signal support ALSA: pcm: Drop async signal support ALSA: control: Drop async signal support ALSA: core: Drop async signal support ALSA: doc: Drop stale fasync entry .../kernel-api/writing-an-alsa-driver.rst | 1 - include/sound/control.h | 1 - include/sound/pcm.h | 1 - sound/core/control.c | 11 ----------- sound/core/init.c | 11 +---------- sound/core/pcm_lib.c | 4 +--- sound/core/pcm_native.c | 18 ------------------ sound/core/timer.c | 13 ------------- 8 files changed, 2 insertions(+), 58 deletions(-) -- 2.35.3

2 7

[PATCH 0/4] ASoC: SOF: Intel: add support for SoundWire-based HP Omen16
by Pierre-Louis Bossart 15 Jul '22

15 Jul '22

This device exposes a headset codec on link0 and an amplifier on link3. This is a very unusual pin-muxing, usually the microphones are pin-muxed with link2/link3. This resulted in a problematic error handling leading to a kernel oops, and invalidated a hard-coded assumption. Full support for this device requires a DMI quirk shared separately ("soundwire: dmi-quirks: add remapping for HP Omen 16-k0005TX"). Pierre-Louis Bossart (4): ASoC: Intel: sof_sdw: avoid oops in error handling ASoC: Intel: soc-acpi: add table for HP Omen 16-k0005TX ASoC: Intel: sof_sdw: add quirk for HP Omen 16-k0005TX ASoC: SOF: Intel: enable dmic handling with 2 or fewer SoundWire links sound/soc/intel/boards/sof_sdw.c | 9 ++++++ sound/soc/intel/boards/sof_sdw_rt711.c | 3 ++ sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 3 ++ .../intel/common/soc-acpi-intel-adl-match.c | 29 +++++++++++++++++++ sound/soc/sof/intel/hda.c | 8 ++--- 5 files changed, 48 insertions(+), 4 deletions(-) -- 2.34.1

2 5

[PATCH] ASoC: codecs: add support for ES8326
by Zhu Ning 15 Jul '22

15 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 794 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 +++++++++ 4 files changed, 988 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..d62c596ac878 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,794 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS_1B, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS_1B, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_RATES SNDRV_PCM_RATE_8000_96000 + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DETECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + es8326_disable_micbias(comp); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if(!es8326->jack) + goto out; + es8326_enable_micbias(comp); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_OFF); + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) + snd_soc_jack_report(es8326->jack, 0, SND_JACK_BTN_0); + + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..9affd5c14407 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.c -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

3 2

[PATCH 0/4] ASoC: Intel: Mark BE DAIs as nonatomic for hsw and
by Cezary Rojewski 15 Jul '22

15 Jul '22

Address the warning: "Codec: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic" by marking BE DAI as nonatomic. Aligns with what is already done for FE DAIs. This patchset iterates the change over all HSW and BDW related machine board drivers. Cezary Rojewski (4): ASoC: Intel: hsw_rt5640: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt286: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5650: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5677: Mark BE DAI as nonatomic sound/soc/intel/boards/bdw-rt5650.c | 1 + sound/soc/intel/boards/bdw-rt5677.c | 1 + sound/soc/intel/boards/bdw_rt286.c | 1 + sound/soc/intel/boards/hsw_rt5640.c | 1 + 4 files changed, 4 insertions(+) -- 2.25.1

3 13

[PATCH] Revert "ALSA: hda: Fix page fault in snd_hda_codec_shutdown()"
by Takashi Iwai 15 Jul '22

15 Jul '22

This reverts commit 980b3a8790b402e959a6d773b38b771019682be1. The commit didn't consider the fact that ASoC hdac-hda driver initializes the HD-audio stuff without calling snd_hda_codec_device_init(). Hence this caused a regression leading to Oops. Revert the commit to restore the behavior. Fixes: 980b3a8790b4 ("ALSA: hda: Fix page fault in snd_hda_codec_shutdown()") Link: https://lore.kernel.org/r/3c40df55-3aee-1e08-493b-7b30cd84dc00@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7be74227bf19..7b2e62fa82d5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,28 +931,8 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, } codec->bus = bus; - codec->depop_delay = -1; - codec->fixup_id = HDA_FIXUP_ID_NOT_SET; - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; - mutex_init(&codec->spdif_mutex); - mutex_init(&codec->control_mutex); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); - snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); - snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); - INIT_LIST_HEAD(&codec->conn_list); - INIT_LIST_HEAD(&codec->pcm_list_head); - INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); - refcount_set(&codec->pcm_ref, 1); - init_waitqueue_head(&codec->remove_sleep); - return codec; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_init); @@ -1005,8 +985,29 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.exec_verb = codec_exec_verb; + codec->card = card; codec->addr = codec_addr; + mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); + snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); + INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); + + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM codec->power_jiffies = jiffies; -- 2.35.3

1 0

[PATCH] Revert "ALSA: hda: cs35l41: Allow compilation test on non-ACPI configurations"
by Takashi Iwai 15 Jul '22

15 Jul '22

Since the recent change in CS35L41 codec requires the reference of acpi_dev handle, the current Kconfig may lead to a build breakage. Revert the Kconfig change and re-introduce the hard dependency on CONFIG_ACPI again as a temporary workaround. Fixes: eef375960210 ("ALSA: hda: cs35l41: Support reading subsystem id from ACPI") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 44c33bc0740e..a8e8cf98befa 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -103,7 +103,7 @@ config SND_HDA_CS_DSP_CONTROLS config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on I2C - depends on ACPI || COMPILE_TEST + depends on ACPI depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 @@ -118,7 +118,7 @@ comment "Set to Y if you want auto-loading the side codec driver" config SND_HDA_SCODEC_CS35L41_SPI tristate "Build CS35L41 HD-audio codec support for SPI Bus" depends on SPI_MASTER - depends on ACPI || COMPILE_TEST + depends on ACPI depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 -- 2.35.3

1 0

[PATCH v6 0/2] Read _SUB from ACPI to be able to identify firmware
by Stefan Binding 15 Jul '22

15 Jul '22

CS35L41 has a DSP which is able to run firmware, as well as a tuning file. Different systems may want to use different firmwares and tuning files, and some firmwares/tunings may not be compatible with other systems. To allow a system to select the correct fimware/tuning, we can read an _SUB from the ACPI. This _SUB can then be used to uniquely identify the system in the firmware/tuning file name. Add a helper function which reads the _SUB, so this can be used by other parts in the future. Add support inside the CS35L41 ASoC driver to read this _SUB, and save it appropriately. Changes since v5: - Clean up return codes - Refactor length calculation - Allow fallback to existing behaviour when bad ACPI is detected Changes since v4: - Rename function Changes since v3: - Fix 32 bit format string warning Changes since v2: - Fix error in function prototype Changes since v1: - Add length validation for SSID String - Rename API - Allocate memory inside API - Use ACPI_HANDLE macro instead of ACPI_COMPANION - Improve error handling Stefan Binding (2): ACPI: utils: Add api to read _SUB from ACPI ASoC: cs35l41: Read System Name from ACPI _SUB to identify firmware drivers/acpi/utils.c | 38 ++++++++++++++++++++++++++++++++++++++ include/linux/acpi.h | 6 ++++++ sound/soc/codecs/cs35l41.c | 30 ++++++++++++++++++++++++++++++ 3 files changed, 74 insertions(+) -- 2.25.1

3 4

[PATCH] soundwire: dmi-quirks: add remapping for HP Omen 16-k0005TX
by Pierre-Louis Bossart 15 Jul '22

15 Jul '22

The DSDT for this device has a number of problems: a) it lists rt711 on link0 and link1, but link1 is disabled b) the rt711 entry on link0 uses the wrong v2 instead of v3 (SDCA) c) the rt1316 amplifier on link3 is not listed. Add a remapping table to work-around these BIOS shenanigans. BugLink: https://github.com/thesofproject/sof/issues/5955 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/dmi-quirks.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 747983743a14..f81cdd83ec26 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -55,7 +55,26 @@ static const struct adr_remap dell_sku_0A3E[] = { {} }; +/* + * The HP Omen 16-k0005TX does not expose the correct version of RT711 on link0 + * and does not expose a RT1316 on link3 + */ +static const struct adr_remap hp_omen_16[] = { + /* rt711-sdca on link0 */ + { + 0x000020025d071100ull, + 0x000030025d071101ull + }, + /* rt1316-sdca on link3 */ + { + 0x000120025d071100ull, + 0x000330025d131601ull + }, + {} +}; + static const struct dmi_system_id adr_remap_quirk_table[] = { + /* TGL devices */ { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "HP"), @@ -78,6 +97,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)dell_sku_0A3E, }, + /* ADL devices */ + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "OMEN by HP Gaming Laptop 16-k0xxx"), + }, + .driver_data = (void *)hp_omen_16, + }, {} }; -- 2.34.1

1 0

[PATCH v8 00/14] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 15 Jul '22

15 Jul '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. changes since v7: - Use private_data rather than private_value to save control info - Clean up alsa control memory allocation/deallocation - Remove unnecessary whitespace - Get subsystem id from codec, rather than saving it separately changes since v6: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v5: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v4: - Fully remove tlv remnants from control add apis - Remove unnecessary debug - Rename variable to be more generic - Remove redundent length check from read/write control apis - Use SNDRV_CTL_ELEM_IFACE_CARD for firmware load controls - Make kcontrol add/remove synchronous - Load firmware asynchronous when loading via control - Used cached controls when reloading firmware; only delete controls when removing the driver itself - Improve kcontrol remove - Fix control write + notify - Cleanup of unnecessary code - Fix race condition when loading firmware before playback - Ensure errors are properly propogated - Fix include for Module parameters Stefan Binding (13): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 4 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 883 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 39 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 3 + sound/pci/hda/hda_cs_dsp_ctl.c | 240 +++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 39 ++ sound/pci/hda/patch_realtek.c | 26 +- 12 files changed, 1242 insertions(+), 5 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

2 22

[GIT PULL] ASoC updates for v5.20
by Mark Brown 15 Jul '22

15 Jul '22

The following changes since commit fc976f5629afb4160ee77798b14a693eac903ffd: ASoC: Intel: Skylake: Correct the handling of fmt_config flexible array (2022-07-07 17:16:40 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.20 for you to fetch changes up to 7fb72b7bf167a8047204d30e0e8affe6023363d9: ASoC: Merge up revert of v5.19 version of the rockchip BCLK management (2022-07-14 16:42:29 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.20 This is a big release thus far and there will probably be more changes to come, it's a combination of a larger than usual crop of new drivers and some subsysetm wide cleanups from Charles rather than anything structural. The SOF and Intel DSP code both also continue to be very actively developed. - Restructing of the set_fmt() callbacks to be specified in terms of the device rather than with semantics depending on if the device is supposed to be a CODEC or SoC, making things clearer in situations like CODEC to CODEC links. - Clean up of the way we flag which DAI naming scheme we use to reflect the progress that's been made modernising things. - Merge of more of the Intel AVS driver stack, including some board integrations. - New version 4 mechanism for communication with SOF DSPs. - Suppoort for dynamically selecting the PLL to use at runtime on i.MX platforms. - Improvements for CODEC to CODEC support in the generic cards. - Support for AMD Jadeite and various machines, Intel MetorLake DSPs, Mediatek MT8186 DSPs and MT6366, nVidia Tegra MDDRC, OPE and PEQ, NXP TFA9890, Qualcomm SDM845, WCD9335 and WAS883x, and Texas Instruments TAS2780. ---------------------------------------------------------------- Aidan MacDonald (2): ASoC: jz4740-i2s: Remove Open Firmware dependency ASoC: jz4740-i2s: Refactor DMA channel setup Ajit Kumar Pandey (2): ASoC: SOF: amd: Add SOF pm ops callback for Renoir ASoC: SOF: topology: add code to parse config params for ACPDMIC dai Alexander Martinz (2): ASoC: codecs: tfa989x: Add support for tfa9890 ASoC: dt-bindings: nxp,tfa989x: Add tfa9890 support Alexandru Elisei (1): ASoC: rockchip: i2s: Fix NULL pointer dereference when pinctrl is not found Amadeusz Sławiński (10): ASoC: Intel: avs: Add max98373 machine board ASoC: Intel: avs: Fix parsing UUIDs in topology ASoC: Remove unused hw_write_t type ASoC: Intel: avs: Set max DMA segment size ASoC: Intel: avs: Use helper function to set up DMA ASoC: codecs: rt298: Fix NULL jack in interrupt ASoC: codecs: rt298: Fix jack detection ASoC: codecs: rt286: Set component to NULL on remove ASoC: codecs: rt298: Set component to NULL on remove ASoC: codecs: rt274: Set component to NULL on remove Andy Shevchenko (4): ASoC: madera: Replace kernel.h with the necessary inclusions ASoC: Intel: catpt: remove duplicating driver data retrieval ASoC: SOF: Intel: bdw: remove duplicating driver data retrieval ASoC: SOF: Intel: byt: remove duplicating driver data retrieval Arnaud Ferraris (2): ASoC: dt-bindings: sun50i-codec: Add binding for internal bias ASoC: sun50i-codec-analog: Add support for internal bias Bard Liao (7): ASoC: SOF: Intel: add MeteorLake machines ASoC: SOF: IPC4: add sdw blob ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support ASoC: SOC: Intel: introduce cl_init callback ASoC: SOF: Intel: Add support for MeteorLake (MTL) soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback Brent Lu (2): ASoC: Intel: sof_cs42l42: support BT offload audio ASoC: Intel: sof_cs42l42: add adl_mx98360a_cs4242 board config Bryan O'Donoghue (1): ASoC: qcom: lpass: Fix apq8016 compat string to match yaml Cezary Rojewski (43): ALSA: Add snd_pcm_direction_name() helper ASoC: codecs: Add HD-Audio codec driver ASoC: Intel: avs: Add HDAudio machine board ASoC: Intel: avs: Add DMIC machine board ASoC: Intel: avs: Add I2S-test machine board ASoC: Intel: avs: Add rt274 machine board ASoC: Intel: avs: Add rt286 machine board ASoC: Intel: avs: Add rt298 machine board ASoC: Intel: avs: Add rt5682 machine board ASoC: Intel: avs: Add nau8825 machine board ASoC: Intel: avs: Add ssm4567 machine board ASoC: Intel: avs: Add max98357a machine board ASoC: Intel: avs: Add da7219 machine board ASoC: codecs: rt274: Always init jack_detect_work ASoC: codecs: rt286: Reorganize jack detect handling ASoC: codecs: rt298: Reorganize jack detect handling ASoC: Intel: Rename haswell source file to hsw_rt5640 ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members ASoC: Intel: hsw_rt5640: Reword driver name ASoC: Intel: hsw_rt5640: Update code indentation ASoC: Intel: hsw_rt5640: Update file comments ASoC: Intel: hsw_rt5640: Improve probe() function quality ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability ASoC: Intel: Rename broadwell source file to bdw_rt286 ASoC: Intel: bdw_rt286: Reword prefixes of all driver members ASoC: Intel: bdw_rt286: Reword driver name ASoC: Intel: bdw_rt286: Update code indentation ASoC: Intel: bdw_rt286: Update file comments ASoC: Intel: bdw_rt286: Improve probe() function quality ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability ASoC: Intel: bdw_rt286: Improve codec_init() quality ASoC: Intel: bdw_rt286: Remove FE DAI ops ASoC: Intel: avs: Fix i2s_test card name initialization ASoC: Intel: avs: Register HDAudio ext-bus operations ASoC: Intel: avs: Assign I2S gateway when parsing topology ASoC: Intel: avs: Relax DSP core transition timings ASoC: Intel: avs: Copy only as many RX bytes as necessary ASoC: Intel: avs: Shield LARGE_CONFIG_GETs against zero payload_size ASoC: Intel: avs: Block IPC channel on suspend ASoC: Intel: avs: Recognize FW_CFG_RESERVED ASoC: Intel: avs: Replace hardcodes with SD_CTL_STREAM_RESET ASoC: Intel: avs: Lower UNLOAD_MULTIPLE_MODULES IPC timeout ASoC: Intel: avs: Update AVS_FW_INIT_TIMEOUT_US declaration Charles Keepax (168): ASoC: core: Add set_fmt_new callback that directly specifies provider ASoC: amd: vangogh: Update to use set_fmt_new callback ASoC: atmel: Update to use set_fmt_new callback ASoC: au1x: Update to use set_fmt_new callback ASoC: bcm: Update to use set_fmt_new callback ASoC: ep93xx: Update to use set_fmt_new callback ASoC: dwc: Update to use set_fmt_new callback ASoC: fsl: Update to use set_fmt_new callback ASoC: hisilicon: Update to use set_fmt_new callback ASoC: img: Update to use set_fmt_new callback ASoC: Intel: Update to use set_fmt_new callback ASoC: js4740-i2s: Update to use set_fmt_new callback ASoC: mediatek: Update to use set_fmt_new callback ASoC: meson: Update to use set_fmt_new callback ASoC: mxs-saif: Update to use set_fmt_new callback ASoC: pxa: Update to use set_fmt_new callback ASoC: qcom: Update to use set_fmt_new callback ASoC: rockchip: Update to use set_fmt_new callback ASoC: samsung: Update to use set_fmt_new callback ASoC: sh: Update to use set_fmt_new callback ASoC: stm: Update to use set_fmt_new callback ASoC: sunxi: Update to use set_fmt_new callback ASoC: tegra: Update to use set_fmt_new callback ASoC: test-component: Update to use set_fmt_new callback ASoC: ti: Update to use set_fmt_new callback ASoC: ux500: Update to use set_fmt_new callback ASoC: xtensa: Update to use set_fmt_new callback ASoC: core: Always send the CPU DAI a direct clock specifier ASoC: amd: vangogh: Rename set_fmt_new back to set_fmt ASoC: atmel: Rename set_fmt_new back to set_fmt ASoC: au1x: Rename set_fmt_new back to set_fmt ASoC: bcm: Rename set_fmt_new back to set_fmt ASoC: ep93xx: Rename set_fmt_new back to set_fmt ASoC: dwc: Rename set_fmt_new back to set_fmt ASoC: fsl: Rename set_fmt_new back to set_fmt ASoC: hisilicon: Rename set_fmt_new back to set_fmt ASoC: img: Rename set_fmt_new back to set_fmt ASoC: Intel: Rename set_fmt_new back to set_fmt ASoC: jz4740-i2s: Rename set_fmt_new back to set_fmt ASoC: mediatek: Rename set_fmt_new back to set_fmt ASoC: meson: Rename set_fmt_new back to set_fmt ASoC: mxs-saif: Rename set_fmt_new back to set_fmt ASoC: pxa: Rename set_fmt_new back to set_fmt ASoC: qcom: Rename set_fmt_new back to set_fmt ASoC: rockchip: Rename set_fmt_new back to set_fmt ASoC: samsung: Rename set_fmt_new back to set_fmt ASoC: sh: Rename set_fmt_new back to set_fmt ASoC: stm: Rename set_fmt_new back to set_fmt ASoC: sunxi: Rename set_fmt_new back to set_fmt ASoC: tegra: Rename set_fmt_new back to set_fmt ASoC: test-component: Rename set_fmt_new back to set_fmt ASoC: ti: Rename set_fmt_new back to set_fmt ASoC: ux500: Rename set_fmt_new back to set_fmt ASoC: xtensa: Rename set_fmt_new back to set_fmt ASoC: soc-dai: Remove set_fmt_new callback ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local ASoC: cs42l52: Fix TLV scales for mixer controls ASoC: cs35l36: Update digital volume TLV ASoC: cs53l30: Correct number of volume levels on SX controls ASoC: cs42l52: Correct TLV for Bypass Volume ASoC: cs42l56: Correct typo in minimum level for SX volume controls ASoC: cs42l51: Correct minimum value for SX volume control ASoC: sun8i-codec: Partial revert to fix clock specifiers ASoC: mediatek: mt8186: Use new direct clock defines ASoC: cs35l45: Add endianness flag in snd_soc_component_driver ASoC: wm_adsp: Fix event for preloader ASoC: wm5110: Fix DRE control ASoC: cs35l41: Correct some control names ASoC: core: Add new SOC_DOUBLE_SX_TLV macro ASoC: dapm: Move stereo autodisable check ASoC: soc-component: Add legacy_dai_naming flag soundwire: intel: Migrate to new style legacy DAI naming flag drm/vc4: Migrate to new style legacy DAI naming flag ASoC: img: Migrate to new style legacy DAI naming flag ASoC: spear: Migrate to new style legacy DAI naming flag ASoC: jz4740-i2c: Migrate to new style legacy DAI naming flag ASoC: ep93xx: Migrate to new style legacy DAI naming flag ASoC: stm32: Migrate to new style legacy DAI naming flag ASoC: bcm: Migrate to new style legacy DAI naming flag ASoC: sh: Migrate to new style legacy DAI naming flag ASoC: tegra: Migrate to new style legacy DAI naming flag ASoC: hisilicon: Migrate to new style legacy DAI naming flag ASoC: xilinx: Migrate to new style legacy DAI naming flag ASoC: sunxi: Migrate to new style legacy DAI naming flag ASoC: Intel: Migrate to new style legacy DAI naming flag ASoC: meson: Migrate to new style legacy DAI naming flag ASoC: sti-uniperf: Migrate to new style legacy DAI naming flag ASoC: amd: Migrate to new style legacy DAI naming flag ASoC: atmel: Migrate to new style legacy DAI naming flag ASoC: fsl: Migrate to new style legacy DAI naming flag ASoC: xtensa: Migrate to new style legacy DAI naming flag ASoC: adi: Migrate to new style legacy DAI naming flag ASoC: dwc: Migrate to new style legacy DAI naming flag ASoC: qcom: Migrate to new style legacy DAI naming flag ASoC: test-component: Migrate to new style legacy DAI naming flag ASoC: rockchip: Migrate to new style legacy DAI naming flag ASoC: au1x: Migrate to new style legacy DAI naming flag ASoC: pxa: Migrate to new style legacy DAI naming flag ASoC: sof: Migrate to new style legacy DAI naming flag ASoC: ux500: Migrate to new style legacy DAI naming flag ASoC: ti: Migrate to new style legacy DAI naming flag ASoC: mxs-saif: Migrate to new style legacy DAI naming flag ASoC: samsung: Migrate to new style legacy DAI naming flag ASoC: core: Switch core to new DAI naming flag media: TDA1997x: Remove now redundant non_legacy_dai_naming flag ASoC: fsl: Remove now redundant non_legacy_dai_naming flag ASoC: meson: Remove now redundant non_legacy_dai_naming flag ASoC: pistachio: Remove now redundant non_legacy_dai_naming flag ASoC: samsung: Remove now redundant non_legacy_dai_naming flag ASoC: soc-utils: Remove now redundant non_legacy_dai_naming flag ASoC: sunxi: Remove now redundant non_legacy_dai_naming flag ASoC: tegra: Remove now redundant non_legacy_dai_naming flag ASoC: test-component: Remove now redundant non_legacy_dai_naming flag ASoC: topology: KUnit: Remove now redundant non_legacy_dai_naming flag ASoC: uniphier: Remove now redundant non_legacy_dai_naming flag ASoC: ad*: Remove now redundant non_legacy_dai_naming flag ASoC: ak*: Remove now redundant non_legacy_dai_naming flag ASoC: alc*: Remove now redundant non_legacy_dai_naming flag ASoC: cs*: Remove now redundant non_legacy_dai_naming flag ASoC: da*: Remove now redundant non_legacy_dai_naming flag ASoC: es*: Remove now redundant non_legacy_dai_naming flag ASoC: max*: Remove now redundant non_legacy_dai_naming flag ASoC: msm*: Remove now redundant non_legacy_dai_naming flag ASoC: nau*: Remove now redundant non_legacy_dai_naming flag ASoC: pcm*: Remove now redundant non_legacy_dai_naming flag ASoC: rt*: Remove now redundant non_legacy_dai_naming flag ASoC: spdif: Remove now redundant non_legacy_dai_naming flag ASoC: ssm*: Remove now redundant non_legacy_dai_naming flag ASoC: sta*: Remove now redundant non_legacy_dai_naming flag ASoC: tas*: Remove now redundant non_legacy_dai_naming flag ASoC: tfa*: Remove now redundant non_legacy_dai_naming flag ASoC: tlv320*: Remove now redundant non_legacy_dai_naming flag ASoC: twl*: Remove now redundant non_legacy_dai_naming flag ASoC: uda*: Remove now redundant non_legacy_dai_naming flag ASoC: wm*: Remove now redundant non_legacy_dai_naming flag ASoC: 88pm860x: Remove now redundant non_legacy_dai_naming flag ASoC: ab8500: Remove now redundant non_legacy_dai_naming flag ASoC: ac97: Remove now redundant non_legacy_dai_naming flag ASoC: ads117x: Remove now redundant non_legacy_dai_naming flag ASoC: bd28623: Remove now redundant non_legacy_dai_naming flag ASoC: bt-sco: Remove now redundant non_legacy_dai_naming flag ASoC: cpcap: Remove now redundant non_legacy_dai_naming flag ASoC: cq93vc: Remove now redundant non_legacy_dai_naming flag ASoC: cx20442: Remove now redundant non_legacy_dai_naming flag ASoC: dmic: Remove now redundant non_legacy_dai_naming flag ASoC: gtm601: Remove now redundant non_legacy_dai_naming flag ASoC: hdac_hdmi: Remove now redundant non_legacy_dai_naming flag ASoC: hdmi-codec: Remove now redundant non_legacy_dai_naming flag ASoC: ics43432: Remove now redundant non_legacy_dai_naming flag ASoC: inno_rk3036: Remove now redundant non_legacy_dai_naming flag ASoC: Intel: avs: Remove now redundant non_legacy_dai_naming flag ASoC: isabelle: Remove now redundant non_legacy_dai_naming flag ASoC: jz4740: Remove now redundant non_legacy_dai_naming flag ASoC: lm49453: Remove now redundant non_legacy_dai_naming flag ASoC: lochnagar: Remove now redundant non_legacy_dai_naming flag ASoC: mc13783: Remove now redundant non_legacy_dai_naming flag ASoC: ml26124: Remove now redundant non_legacy_dai_naming flag ASoC: rk817: Remove now redundant non_legacy_dai_naming flag ASoC: sgtl5000: Remove now redundant non_legacy_dai_naming flag ASoC: si476x: Remove now redundant non_legacy_dai_naming flag ASoC: stac9766: Remove now redundant non_legacy_dai_naming flag ASoC: sti-sas: Remove now redundant non_legacy_dai_naming flag ASoC: tscs42xx: Remove now redundant non_legacy_dai_naming flag ASoC: wl1273: Remove now redundant non_legacy_dai_naming flag ASoC: zl38060: Remove now redundant non_legacy_dai_naming flag ASoC: soc-component: Remove non_legacy_dai_naming flag ASoC: samsung: s3c24xx-i2s: Fix typo in DAIFMT handling ASoC: wm_adsp: Minor clean and redundant code removal Chen-Yu Tsai (1): ASoC: rockchip: i2s: Fix crash on missing pinctrl Christophe JAILLET (2): ASoC: ux500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules ASoC: ab8500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules Claudiu Beznea (1): ASoC: atmel: mchp-pdmc: remove space in front of mchp_pdmc_dt_init() Colin Ian King (2): ASoC: mediatek: mt8186: Fix a handful of spelling mistakes ASoC: Intel: Skylake: remove redundant re-assignments to pointer array Conor Dooley (1): ASoC: dt-bindings: convert designware-i2s to dt-schema Cristian Ciocaltea (1): ASoC: cs35l41: Add support for CLSA3541 ACPI device ID Dan Carpenter (4): ASoC: SOF: mediatek: Fix error code in probe ASoC: SOF: ipc4-topology: Fix error code in sof_ipc4_volume_put() ASoC: amd: Fix error pointer dereference ASoC: SOF: ipc-msg-injector: fix copy in sof_msg_inject_ipc4_dfs_write() Daniel Baluta (4): ASoC: SOF: compress: Dynamically allocate pcm params struct ASoC: SOF: Copy compress parameters into extended data ASoC: SOF: compress: Prevent current kernel running with older FW uapi: sof: abi: Bump SOF ABI for ext_data_length Daniel Mack (7): ASoC: dt-bindings: max98396: add voltage supplies ASoC: dt-bindings: max98396: Document adi,bypass-slot-no ASoC: max98396: add voltage regulators ASoC: max98396: Improve some error prints ASoC: max98396: Fix register access for PCM format settings ASoC: max98396: Implement DSP speaker monitor ASoC: max98396: Fix TDM mode BSEL settings David Lin (1): ASoC: nau8825: Declare 2 channels for DAI of capture stream Fabio Estevam (1): ASoC: imx-audmux: Silence a clang warning Fei Shao (1): ASoC: mediatek: mt8186: Fix mutex double unlock in GPIO request Francesco Dolcini (1): ASoC: sgtl5000: Fix noise on shutdown/remove Gaosheng Cui (1): ASoC: codecs: wsa883x: fix warning using-module-alias-sdw.cocci Hans de Goede (1): ASoC: Intel: bytcr_wm5102: Fix GPIO related probe-ordering problem Hui Wang (2): ASoC: nau8822: Disable internal PLL if freq_out is zero ASoC: nau8822: Add operation for internal PLL off and on Javier Martinez Canillas (6): regmap: Re-introduce bulk read support check in regmap_bulk_read() regmap: Make regmap_noinc_read() return -ENOTSUPP if map->read isn't set regmap: Wire up regmap_config provided bulk write in missed functions regmap: Re-introduce bulk read support check in regmap_bulk_read() regmap: Make regmap_noinc_read() return -ENOTSUPP if map->read isn't set regmap: Wire up regmap_config provided bulk write in missed functions Jiapeng Chong (3): ASoC: rockchip: i2s: Fix missing error code in rockchip_i2s_probe() ASoC: codecs: max98088: Clean up some inconsistent indenting ASoC: tegra20_ac97: Fix missing error code in tegra20_ac97_platform_probe() Jiasheng Jiang (1): ASoC: codecs: da7210: add check for i2c_add_driver Jiaxin Yu (12): ASoC: mediatek: mt6366: support for mt6366 codec ASoC: mediatek: mt8186: support audsys clock control ASoC: mediatek: mt8186: support adda in platform driver ASoC: mediatek: mt8186: support hostless in platform driver ASoC: mediatek: mt8186: support hw gain in platform driver ASoC: mediatek: mt8186: support i2s in platform driver ASoC: mediatek: mt8186: support pcm in platform driver ASoC: mediatek: mt8186: support src in platform driver ASoC: mediatek: mt8186: support tdm in platform driver ASoC: mediatek: mt8186: support audio clock control in platform driver ASoC: mediatek: mt8186: support gpio control in platform driver ASoC: mediatek: mt8186: add misc driver and register definitions Judy Hsiao (6): ASoC: rockchip: i2s: switch BCLK to GPIO ASoC: rockchip: i2s: switch BCLK to GPIO ASoC: dt-bindings: rockchip: Document pinctrl-names for i2s ASoC: rockchip: i2s: Fix the debug level on missing pinctrl ASoC: rockchip: i2s: Remove unwanted dma settings in rockchip_i2s_probe ASoC: rockchip: i2s: Fix error code when fail to read I2S_CLR Julia Lawall (2): ASoC: amd: acp: fix typo in comment ASoC: stm32: dfsdm: fix typo in comment Krzysztof Kozlowski (5): ASoC: samsung: s3c-i2s-v2: Allow build for unsupported hardware ASoC: samsung: s3c24xx-i2s: Drop unneeded gpio.h include ASoC: samsung: Enable compile test ASoC: samsung: h1940_uda1380: include proepr GPIO consumer header ASoC: dt-bindings: qcom,sm8250: add SDM845 sound Kuninori Morimoto (11): ASoC: simple-card-utils: rename asoc_simple_init_dai_link_params() to asoc_simple_init_for_codec2codec() ASoC: ak4613: cares Simple-Audio-Card case for TDM ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus() ASoC: soc-core.c: share code for snd_soc_of_get_dai_link_cpus/codecs() ASoC: audio-graph-card2.c: use of_property_read_u32() for rate ASoC: audio-graph-card2.c: make Codec2Codec settings optional ASoC: audio-graph-card2.c: remove pre-alloced Codec2Codec space ASoC: audio-graph-card2-custom-sample.dtsi: add verbose explanation ASoC: simple-card-utils.c: ignore Codec2Codec setting if it already have ASoC: simple-card-utils.c: care Codec2Codec vs DPCM:BE ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl() Li Chen (2): regmap: provide regmap_field helpers for simple bit operations ASoC: sunxi: Use {regmap/regmap_field}_{set/clear}_bits helpers Li kunyu (1): ASoC: tegra: delete a semicolon Liang He (3): ASoC: qcom: Fix missing of_node_put() in asoc_qcom_lpass_cpu_platform_probe() ASoc: audio-graph-card2: Fix refcount leak bug in __graph_get_type() ASoC: mt6359: Fix refcount leak bug Libin Yang (2): ASoC: SOF: ipc4-topology: check dai->private in ipc_free() ASoC: SOF: ipc4-topology: free memories allocated in sof_ipc4_get_audio_fmt Lukas Bulwahn (1): ASoC: Intel: avs: correct config reference for I2S test board Marco Felsch (2): ASoC: fsl_sai: use local device pointer ASoC: fsl_sai: add error message in case of missing imx-pcm-dma support Marek Vasut (1): ASoC: doc: Update dead links Mark Brown (93): Merge existing fixes from asoc/for-5.19 into new branch ASoC: ssm2518: Use modern ASoC DAI format terminology ASoC: ssm2602: Use modern ASoC DAI format terminology ASoC: ssm4567: Use modern ASoC DAI format terminology ASoC: ops: Clarify snd_soc_info_volsw_sx() ASoC: nau8822: Don't reconfigure PLL to the same values ASoC: ops: Remove unneeded delay.h inclusion ASoC: tfa9879: Use modern ASoC DAI format terminology ASoC: max9860: Use modern ASoC DAI format terminology ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put() ASoC: cx2072x: Use modern ASoC DAI format terminology ASoC: max98090: Use modern ASoC DAI format terminology ASoC: rk3328: Use modern ASoC DAI format terminology ASoC: sta32x: Use modern ASoC DAI format terminology ASoC: sta350: Use modern ASoC DAI format terminology ASoC: sti-sas: Use modern ASoC DAI format terminology ASoC: tas2552: Use modern ASoC DAI format terminology ASoC: tas2770: Use modern ASoC DAI format terminology ASoC: tas5086: Use modern ASoC DAI format terminology ASoC: tas5720: Use modern ASoC DAI format terminology ASoC: tas6424: Use modern ASoC DAI format terminology ASoC: uda1334: Use modern ASoC DAI format terminology ASoC: tlv320adc3xxx: Use modern ASoC DAI format terminology ASoC: tlv320adcx140: Use modern ASoC DAI format terminology ASoC: tlv320aic23: Use modern ASoC DAI format terminology ASoC: tlv320aic26: Use modern ASoC DAI format terminology ASoC: tlv320aic31xx: Use modern ASoC DAI format terminology ASoC: tlv320aic32x4: Use modern ASoC DAI format terminology ASoC: tlv320aic33: Use modern ASoC DAI format terminology ASoC: tlv320dac3x: Use modern ASoC DAI format terminology ALSA: hda: cirrus: Add initial DSP support and firmware loading ASoC: Intel: soc-acpi and machine driver updates ASoC: Intel: avs: Machine boards and HDA codec support OPE support on Tegra210 and later ASoC: trivial changes for cppcheck warnings Switch to use internal PLL for iMCLK ASoC: Drop some i2c noop remove callbacks Add Machine driver support for nau8825, max98560 and rt5682s, rt1019 ASoC: SOF: AMD/Mediatek updates for 5.20 ASoC: ssm: Use modern ASoC DAI format terminology ASoC: DAI clocking terminology modernisation ASoC: harden SoundWire codec/machine drivers used on Intel platforms Specify clock provider directly to CPU DAIs ASoC: codecs: Series of fixes for realtek codecs used on RVPs ASoC: codecs: qualcomm move gain to S8_TLV ASoC: SOF: ipc3-dtrace: Handle race during initialization ASoC: SOF: Add support ctx_save with IPC4 ASoC: wcd9335: Fix spurious event generation ASoC: hdmi-codec: Update to modern DAI terminology ASoC: wcd938x: Fix event generation for some controls ASoC: SOF: IPC4: Add topology, control and PCM ops ASoC: Merge fixes ASoC: ops: Don't modify the driver's plaform_max when reading state ASoC: ops: Fix boolean/integer detection for simple controls ASoC: SOF: Intel: Add support for MeteorLake ASoC: Add regmap_field helpers for simple bit operations ASoC: SOF/Intel: remove __func__ from dev_dbg() ASoC: minor kernel-doc corrections ASoC: Intel: Chromebooks: remap jack pins ASoC: SOF: Intel: fix resume from hibernate ASoC: SOF/soundwire: use resume_and_get on component probe remap: Some fixes for bulk read/write callbacks in regmap_config support ASoC: Merge fixes Add MT8186 ADSP dt-binding ASoC: rockchip: i2s: switch BCLK to GPIO ASoC: Intel: haswell and broadwell boards update ASoC: Merge up fixes ASoC: sun50i-codec-analog: Internal bias support ASoC: max98396: Some assorted fixes and additions ASoC: use pm_runtime_resume_and_get() when possible Add PDM/DSD/dataline configuration support ASoC: codecs: add WSA883x support ASoC: Refactor non_legacy_dai_naming flag ASoC: dt-bindings: Add bindings for WCD9335 DAIs ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus() Add support of two Audio PLL source I2S driver changes for Jadeite platform ASoC: cleanups and improvements for jz4740-i2s ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload ASoC: audio-graph-card2.c: make Codec2Codec settings optional ASoC: Merge up fixes ASoC: SOF: ipc4-topology: fix error and memory handling ADD legacy audio driver support for rembrandt ASoC: codecs: Series of fixes for realtek codecs used on RVPs ASoC: Intel: avs: Updates and cleanups ASoC: SOF: Intel: hda: Use cold/purge boot after firmware crash ASoC: SOF: Intel: hda-dai: No need to decouple host/link DMA twice ASoC: SOF: Intel: Do not process IPC reply before firmware boot Extend ipc stream parameters sent to DSP ASoC: SOF: Intel: hda: Correct Firmware State Register use ASoC/SoundWire: Intel: add sdw BE dai trigger ASoC: rockchip-i2s: Undo BCLK pinctrl changes ASoC: Merge up revert of v5.19 version of the rockchip BCLK management Martin Povišer (1): ASoC: Introduce 'fixup_controls' card method Mauro Carvalho Chehab (1): ASoC: wm8731: update wlf,wm8731.yaml reference Miaoqian Lin (5): ASoC: cros_ec_codec: Fix refcount leak in cros_ec_codec_platform_probe ASoC: samsung: Fix error handling in aries_audio_probe ASoC: mediatek: mt8173: Fix refcount leak in mt8173_rt5650_rt5676_dev_probe ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe ASoC: mediatek: mt8173-rt5650: Fix refcount leak in mt8173_rt5650_dev_probe Minghao Chi (2): ASoC: mediatek: remove unnecessary check of clk_disable_unprepare ASoC: imx-audmux: remove unnecessary check of clk_disable_unprepare/clk_prepare_enable Oder Chiou (1): ASoC: rt5640: Add the MICBIAS1 to the dapm routing Peter Ujfalusi (27): ASoC: SOF: Intel: hda-dsp: Expose hda_dsp_core_power_up() ASoC: SOF: Intel: hda-loader: Make sure that the fw load sequence is followed ASoC: SOF: Intel: hda-loader: Clarify the cl_dsp_init() flow ASoC: SOF: ipc3-topology: Move and correct size checks in sof_ipc3_control_load_bytes() ASoC: SOF: make ctx_store and ctx_restore as optional ASoC: SOF: sof_ipc_pm_ops: Add support for DSP core power management ASoC: SOF: ipc3: Add set_core_state pm_ops implementation ASoC: SOF: ipc4: Add set_core_state pm_ops implementation ASoC: SOF: Intel: Switch to use the generic pm_ops.set_core_state ASoC: SOF: ipc4: implement pm ctx_save callback ASoC: SOF: ipc3-dtrace: Introduce SOF_DTRACE_INITIALIZING state ASoC: SOF: ipc3-dtrace: Add helper function to update the sdev->host_offset ASoC: SOF: ipc3-dtrace: Return from dtrace_read if there is no new data available ASoC: SOF: debug: Clarify the IPC timeout handling path ASoC: SOF: Compile and runtime IPC version selection ASoC: SOF: Intel: IPC4: enable IMR boot ASoC: SOF: Intel: hda: Fix compressed stream position tracking ASoC: twl4030: Drop legacy, non DT boot support ASoC: SOF: Intel: cnl: Do not process IPC reply before firmware boot ASoC: SOF: Intel: hda-ipc: Do not process IPC reply before firmware boot ASoC: SOF: Intel: mtl: Do not process IPC reply before firmware boot ASoC: SOF: Intel: hda: Introduce skip_imr_boot flag ASoC: SOF: Intel: hda: Correct the ROM/FW state reporting code ASoC: SOF: Intel: hda-dai: Drop misleading comment regarding dma_data ASoC: SOF: Intel: hda-dai: Do snd_hdac_ext_stream_decouple() only once ASoC: SOF: sof-client-probes: Only load the driver if IPC3 is used ASoC: SOF: ipc3-topology: Prevent double freeing of ipc_control_data via load_bytes Pierre-Louis Bossart (74): ASoC: cs35l45: typo in argument definition ASoC: cs42l42: remove redundant test ASoC: wcd-mbhc-v2: remove useless initialization ASoC: wcd9335: remove redundant tests ASoC: Intel: atom: sst: remove useless initialization ASoC: Intel: atom: sst_ipc: remove redundant test ASoC: Intel: atom: sst_ipc: remove useless initializations ASoC: Intel: atom: controls: remove useless initializations ASoC: Intel: boards: reset acpi_chan_package ASoC: Intel: sof_pcm512x: remove unnecessary init ASoC: mediatek: mt8195: simplify error handling ASoC: qcom: q6dsp: q6adm: remove useless initializations ASoC: qcom: q6dsp: remove spurious space ASoC: rockchip: simplify error handling ASoC: samsung: snow: simplify error handling ASoC: meson: remove useless initialization ASoC: Intel: sof_sdw: allow HDaudio/HDMI disable ASoC: Realtek/Maxim SoundWire codecs: disable pm_runtime on remove ASoC: rt711-sdca-sdw: fix calibrate mutex initialization ASoC: Intel: sof_sdw: handle errors on card registration ASoC: rt711: fix calibrate mutex initialization ASoC: rt7*-sdw: harden jack_detect_handler ASoC: codecs: rt700/rt711/rt711-sdca: initialize workqueues in probe ASoC: codecs: rt700/rt711/rt711-sdca: resume bus/codec in .set_jack_detect ASoC: SOF: ipc3-dtrace: use pm_runtime_resume_and_get() MAINTAINERS: update ASoC/Intel/SOF maintainers ASoC: SOF: Intel: hda-dai: enhance debug messages ASoC: SOF: reduce default verbosity of IPC logs ASoC: SOF: pm: add explicit behavior for ACPI S1 and S2 ASoC: SOF: pm: add definitions for S4 and S5 states ASoC: SOF: Intel: disable IMR boot when resuming from ACPI S4 and S5 states ASoC: cs4270: update kernel-doc ASoC: sunxi: sun4i-i2s: update kernel-doc ASoC: Intel: skl_nau88l25_max98357a: remap jack pins ASoC: Intel: skl_nau88l25_ssm4567: remap jack pins ASoC: Intel: kbl_rt5663_max98927: remap jack pins ASoC: Intel: kbl_da7219_max98357a: remap jack pins ASoC: Intel: kbl_da7219_max98927: remap jack pins ASoC: Intel: kbl_rt5663_rt5514_max98927: remap jack pins ASoC: Intel: bxt_da7219_max98357a: remap jack pins ASoC: Intel: glk_rt5682_max98357a: remap jack pins ASoC: Intel: cml_rt1011_rt5682: remap jack pins ASoC: Intel: sof_cs42l42: remap jack pins ASoC: Intel: sof_da7219_max98373: remap jack pins ASoC: Intel: sof_nau8825: remap jack pins ASoC: Intel: sof_rt5682: remap jack pins ASoC: SOF: Intel: hda-dsp: report error on power-up/down ASoC: SOF: Intel: hda-stream: report error on stream not opened ASoC: SOF: Intel: hda-dai: remove use of __func__ in dev_dbg ASoC: SOF: Intel: hda-stream: remove use of __func__ in dev_dbg ASoC: SOF: Intel: mtl: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-dtrace: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-loader: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-topology: remove use of __func__ in dev_dbg ASoC: SOF: ipc4-topology remove use of __func__ in dev_dbg ASoC: SOF: sof-client: remove use of __func__ in dev_dbg ASoC: SOF: ipc4: remove use of __func__ in dev_dbg ASoC: Intel: boards: hda: remove use of __func__ in dev_dbg ASoC: Intel: boards: sof_sdw: remove use of __func__ in dev_dbg ASoC: SOF: pcm: use pm_resume_and_get() on component probe ASoC: codecs: soundwire: call pm_runtime_resume() in component probe ASoC: Intel: catpt: use pm_runtime_resume_and_get() ASoC: Intel: skylake: skl-pcm: use pm_runtime_resume_and_get() ASoC: wcd-mbhc-v2: use pm_runtime_resume_and_get() ASoC: wsa881x: use pm_runtime_resume_and_get() ASoC: rockchip: i2s_tdm: use pm_runtime_resume_and_get() ASoC: fsl: fsl_sai: use pm_runtime_resume_and_get() ASoC: img: img-i2s-out: use pm_runtime_resume_and_get() ASoC: rockchip: pdm: use pm_runtime_resume_and_get() ASoC: tas2552: use pm_runtime_resume_and_get() ASoC: ti: davinci-mcasp: use pm_runtime_resume_and_get() ASoC: amd: acp-es8336: use static variables ASoC: SOF: remove warning on ABI checks ASoC: soc-pcm: demote warnings on non-atomic BE connection Randy Dunlap (2): ASoC: max98390: use linux/gpio/consumer.h to fix build ASoC: amd: fix Jadeite kconfig warning and build errors Ranjani Sridharan (23): ASoC: SOF: Add topology tokens for IPC4 ASoC: SOF: IPC4: Introduce topology ops ASoC: SOF: ipc4-topology: Add support for parsing AIF_IN/AIF_OUT widgets ASoC: SOF: ipc4-topology: Add support for parsing DAI_IN/DAI_OUT widgets ASoC: SOF: ipc4-topology: Add prepare op for AIF type widgets ASoC: SOF: ipc4-topology: Add prepare op for DAI type widgets ASoC: SOF: ipc4-topology: Add support for parsing and preparing pga widgets ASoC: SOF: ipc4-topology: Add support for parsing mixer widgets ASoC: SOF: ipc4-topology: Add control_setup op ASoC: SOF: ipc4-topology: Add control IO ops ASoC: SOF: IPC4: Add pcm ops ASoC: SOF: ipc4-topology: Add widget_setup/widget_free ops ASoC: SOF: ipc4-topology: Add route_setup/route_free ops ASoC: SOF: ipc4-topology: Add the dai_config op ASoC: SOF: ipc4-pcm: Expose sof_ipc4_set_pipeline_state() ASoC: SOF: IPC4: set the BE DAI ops ASoC: SOF: Add ops_free ASoC: SOF: Intel: hda: init NHLT for IPC4 ASoC: SOF: Add two new structures for topology manifest data ASoC: SOF: Add a new IPC op for parsing topology manifest ASoC: SOF: ipc4-topology: Add support for SSP/DMIC DAI's AsoC: SOF: ipc4-topology: Add dai_get_clk op ASoC: SOF: topology: remove unused variable Raphael-Xu (2): ASoC: add tas2780 driver ASoC: add tas2780 DT binding Sameer Pujar (5): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module ASoC: tegra: Fix clock DAI format on Tegra210 ASoC: tegra: Fix MBDRC bypass mode check Sascha Hauer (2): ASoC: fsl_asrc_dma: Use dmaengine_terminate_async() ASoC: audio_graph_card2: Fix port numbers in example Seven Lee (1): ASoC: nau8821: Add headset button detection Shengjiu Wang (25): ASoC: fsl_asrc_dma: enable dual fifo for ASRC P2P ASoC: fsl_mqs: simplify the code with adding fsl_mqs_soc_data ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode ASoC: dt-bindings: fsl,mqs: Add compatible string for i.MX93 platform ASoC: fsl_mqs: Add support for i.MX93 platform ASoC: fsl_sai: Add PDM daifmt support ASoC: fsl_sai: Add DSD bit format support ASoC: fsl_sai: Add support for more sample rates ASoc: fsl_sai: Add pinctrl operation for PDM and DSD ASoC: fsl_sai: Make res a member of struct fsl_sai ASoC: dt-bindings: fsl-sai: Add new property to configure dataline ASoC: fsl_sai: Configure dataline/FIFO information from dts property ASoC: fsl_micfil: change micfil default settings ASoC: dt-bindings: fsl,micfil: Convert format to json-schema ASoC: fsl_micfil: Add legacy_dai_naming flag ASoC: fsl_asrc_dma: Add legacy_dai_naming flag ASoC: fsl_utils: Add function to handle PLL clock source ASoC: fsl_spdif: Add support for PLL switch at runtime. ASoC: fsl_micfil: Add support for PLL switch at runtime ASoC: fsl_sai: Add support for PLL switch at runtime ASoC: dt-bindings: fsl_spdif: Add two PLL clock source ASoC: dt-bindings: fsl-sai: Add two PLL clock source ASoC: imx-card: Fix DSD/PDM mclk frequency ASoC: fsl_utils: Don't use plain integer as NULL pointer ASoC: fsl_utils: Drop usage of __clk_get_name() Shuming Fan (1): ASoC: rt711-sdca: fix kernel NULL pointer dereference when IO error Srinivas Kandagatla (11): ASoC: codecs: msm8916-wcd-digital: move gains from SX_TLV to S8_TLV ASoC: codecs: wcd9335: move gains from SX_TLV to S8_TLV ASoC: qdsp6: q6apm-dai: unprepare stream if its already prepared MAINTAINERS: update ASoC Qualcomm maintainer email-id ASoC: dt-bindings: Add WSA883x bindings ASoC: codecs: add wsa883x amplifier support MAINTAINERS: add ASoC Qualcomm codecs ASoC: codecs: wsa883x: add control, dapm widgets and map ASoC: codecs: wsa881x: handle timeouts in resume path ASoC: codecs: wsa883x: handle timeouts in resume path ASoC: codecs: wsa883x: add missing break statement Srinivasa Rao Mandadapu (3): ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback ASoC: qcom: Add driver support for audioreach solution Stefan Binding (4): ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Add common cs35l41 enter hibernate function ASoC: cs35l41: Do not print error when waking from hibernation ASoC: ops: Fix integer detection for when max possible values > 1 Tang Bin (1): ASoC: stm32: sai: Remove useless define Tinghan Shen (4): dt-bindings: dsp: mediatek: Use meaningful names for mbox firmware: mediatek: Use meaningful names for mbox dt-bindings: dsp: mediatek: Add mt8186 dsp document ASoC: SOF: mediatek: Align mt8186 clock names with dt-bindings Tom Rix (2): ASoC: samsung: change gpiod_speaker_power and rx1950_audio from global to static variables ASoC: samsung: change neo1973_audio from a global to static Uwe Kleine-König (17): ASoC: ak4642: Drop no-op remove function ASoC: da7219: Drop no-op remove function ASoC: lm49453: Drop no-op remove function ASoC: da732x: Drop no-op remove function ASoC: cs35l45: Make cs35l45_remove() return void ASoC: Intel: broadwell: Make broadwell_disable_jack() return void ASoC: simple-card-utils: Make asoc_simple_clean_reference() return void ASoC: core: Make snd_soc_unregister_card() return void ASoC: amd: acp: Fix error handling in .remove() ASoC: topology: KUnit: Followup prototype change of snd_soc_unregister_card() ASoC: xilinx: Suppress second error message about reset failure in .remove() ASoC: tegra: tegra20_das: Fold header file into only user ASoC: tegra: tegra20_das: Remove unused function tegra20_das_read ASoC: tegra: tegra20_das: Get rid of global pointer for driver data ASoC: tegra: tegra20_das: Make helper functions return void ASoC: tegra: tegra20_das: Drop write-only driver data member ASoC: rsnd: Emit useful error messages in .remove() V sujith kumar Reddy (5): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance ASoC: amd: acp: Remove rt1019_1 codec conf from machine driver ASoC: amd: acp: ACP code generic to support newer platforms ASoC: amd: acp: Add legacy audio driver support for Rembrandt platform Vijendar Mukunda (7): ASoC: amd: add I2S MICSP instance support ASoC: amd: add Machine driver for Jadeite platform ASoC: amd: enable machine driver build for Jadeite platform ASoC: amd: fix ACPI dependency compile errors and warnings ASoC: amd: remove unused header file inclusion ASoC: amd: drop machine driver remove function ASoC: amd: fix for variable set but not used warning YC Hung (4): ASoC: SOF: mediatek: revise mt8195 clock sequence ASoC: SOF: mediatek: Add shared_size for mediatek common chip information ASoC: SOF: mediatek: mt8195 modify dram type as non-cache ASoC: SOF: mediatek: mt8195 suspend check dsp idle Yang Li (1): ASoC: amd: Remove duplicated include in acp-es8336.c Yang Yingliang (1): ASoC: fsl: pcm030-audio-fabric: use platform_device_unregsiter() Yassine Oudjana (3): ASoC: wcd9335: Remove RX channel from old list before adding it to a new one ASoC: dt-bindings: Add bindings for WCD9335 DAIs ASoC: wcd9335: Use DT bindings instead of local DAI definitions Yong Zhi (2): ASoC: Intel: boards: rename RTL1019 compatible driver to rt1019p ASoC: intel: sof_sdw: add RT711 SDCA card for MTL platform Zhongjun Tan (1): ASoC: mediatek: mt8186: Remove condition with no effect Zhu Ning (1): ASoC: codes: Add support for ES8316 producer mode xliu (1): ASoC: Intel: cirrus-common: fix incorrect channel mapping zhangqilong (1): ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe .../bindings/dsp/mediatek,mt8186-dsp.yaml | 91 + .../bindings/dsp/mediatek,mt8195-dsp.yaml | 10 +- .../devicetree/bindings/sound/adi,max98396.yaml | 30 +- .../sound/allwinner,sun50i-a64-codec-analog.yaml | 5 + .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 2 +- .../devicetree/bindings/sound/designware-i2s.txt | 35 - .../devicetree/bindings/sound/fsl,micfil.txt | 33 - .../devicetree/bindings/sound/fsl,micfil.yaml | 85 + .../devicetree/bindings/sound/fsl,mqs.txt | 2 +- .../devicetree/bindings/sound/fsl,spdif.yaml | 4 + .../devicetree/bindings/sound/fsl-sai.txt | 11 + .../devicetree/bindings/sound/nau8821.txt | 2 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 + .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + .../devicetree/bindings/sound/nxp,tfa989x.yaml | 1 + .../devicetree/bindings/sound/qcom,sdm845.txt | 91 - .../devicetree/bindings/sound/qcom,sm8250.yaml | 3 + .../devicetree/bindings/sound/qcom,wsa883x.yaml | 74 + .../devicetree/bindings/sound/rockchip-i2s.yaml | 7 + .../bindings/sound/snps,designware-i2s.yaml | 94 + .../devicetree/bindings/sound/tas2780.yaml | 72 + Documentation/process/kernel-docs.rst | 2 +- Documentation/sound/soc/codec.rst | 2 +- Documentation/sound/soc/platform.rst | 2 +- MAINTAINERS | 5 + drivers/base/regmap/regmap.c | 37 +- drivers/firmware/mtk-adsp-ipc.c | 36 +- drivers/gpu/drm/bridge/sii902x.c | 5 +- .../gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 2 +- drivers/gpu/drm/exynos/exynos_hdmi.c | 8 +- drivers/gpu/drm/i2c/tda998x_drv.c | 6 +- drivers/gpu/drm/sti/sti_hdmi.c | 8 +- drivers/gpu/drm/vc4/vc4_hdmi.c | 1 + drivers/media/i2c/tda1997x.c | 1 - drivers/soundwire/intel.c | 14 +- drivers/soundwire/qcom.c | 10 + include/dt-bindings/sound/qcom,wcd9335.h | 15 + include/linux/regmap.h | 37 + include/linux/soundwire/sdw_intel.h | 3 + include/sound/cs35l41.h | 3 + include/sound/hdmi-codec.h | 4 +- include/sound/madera-pdata.h | 2 +- include/sound/pcm.h | 19 +- include/sound/simple_card_utils.h | 5 +- include/sound/soc-acpi-intel-match.h | 2 + include/sound/soc-card.h | 1 + include/sound/soc-component.h | 7 +- include/sound/soc-dai.h | 6 + include/sound/soc.h | 15 +- include/sound/sof.h | 1 + include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai.h | 2 +- include/sound/sof/ipc4/header.h | 8 + include/sound/sof/stream.h | 6 +- include/uapi/sound/sof/abi.h | 4 +- include/uapi/sound/sof/header.h | 30 + include/uapi/sound/sof/tokens.h | 44 + sound/pci/ens1370.c | 2 +- sound/soc/adi/axi-i2s.c | 1 + sound/soc/adi/axi-spdif.c | 1 + sound/soc/amd/Kconfig | 12 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-config.c | 30 + sound/soc/amd/acp-es8336.c | 318 +++ sound/soc/amd/acp-pcm-dma.c | 50 +- sound/soc/amd/acp.h | 13 + sound/soc/amd/acp/Kconfig | 12 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 169 +- sound/soc/amd/acp/acp-legacy-mach.c | 32 + sound/soc/amd/acp/acp-mach-common.c | 301 +- sound/soc/amd/acp/acp-mach.h | 9 + sound/soc/amd/acp/acp-pci.c | 6 + sound/soc/amd/acp/acp-pdm.c | 10 +- sound/soc/amd/acp/acp-platform.c | 47 +- sound/soc/amd/acp/acp-rembrandt.c | 401 +++ sound/soc/amd/acp/acp-renoir.c | 48 +- sound/soc/amd/acp/acp-sof-mach.c | 30 + sound/soc/amd/acp/amd.h | 86 +- sound/soc/amd/acp/chip_offset_byte.h | 40 +- sound/soc/amd/mach-config.h | 1 + sound/soc/amd/raven/acp3x-i2s.c | 3 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 13 +- sound/soc/amd/vangogh/acp5x-i2s.c | 5 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 13 +- sound/soc/atmel/atmel-classd.c | 1 + sound/soc/atmel/atmel-i2s.c | 7 +- sound/soc/atmel/atmel-pdmic.c | 1 + sound/soc/atmel/atmel_ssc_dai.c | 25 +- sound/soc/atmel/mchp-i2s-mcc.c | 11 +- sound/soc/atmel/mchp-pdmc.c | 7 +- sound/soc/atmel/mchp-spdifrx.c | 3 +- sound/soc/atmel/mchp-spdiftx.c | 3 +- sound/soc/atmel/mikroe-proto.c | 4 +- sound/soc/au1x/ac97c.c | 3 +- sound/soc/au1x/i2sc.c | 5 +- sound/soc/au1x/psc-ac97.c | 3 +- sound/soc/au1x/psc-i2s.c | 7 +- sound/soc/bcm/bcm2835-i2s.c | 23 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 1 + sound/soc/bcm/cygnus-ssp.c | 11 +- sound/soc/cirrus/ep93xx-ac97.c | 3 +- sound/soc/cirrus/ep93xx-i2s.c | 11 +- sound/soc/codecs/88pm860x-codec.c | 1 - sound/soc/codecs/Kconfig | 28 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ab8500-codec.c | 3 - sound/soc/codecs/ab8500-codec.h | 2 - sound/soc/codecs/ac97.c | 1 - sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad193x.c | 1 - sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/ad73311.c | 1 - sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761.c | 1 - sound/soc/codecs/adau1781.c | 1 - sound/soc/codecs/adau1977.c | 1 - sound/soc/codecs/adau7002.c | 1 - sound/soc/codecs/adau7118.c | 1 - sound/soc/codecs/adav80x.c | 1 - sound/soc/codecs/ads117x.c | 1 - sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4375.c | 1 - sound/soc/codecs/ak4458.c | 2 - sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4554.c | 1 - sound/soc/codecs/ak4613.c | 7 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/ak5386.c | 1 - sound/soc/codecs/ak5558.c | 2 - sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/alc5632.c | 1 - sound/soc/codecs/bd28623.c | 1 - sound/soc/codecs/bt-sco.c | 1 - sound/soc/codecs/cpcap.c | 1 - sound/soc/codecs/cq93vc.c | 1 - sound/soc/codecs/cros_ec_codec.c | 1 + sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs35l33.c | 1 - sound/soc/codecs/cs35l34.c | 1 - sound/soc/codecs/cs35l35.c | 1 - sound/soc/codecs/cs35l36.c | 1 - sound/soc/codecs/cs35l41-lib.c | 82 +- sound/soc/codecs/cs35l41-spi.c | 1 + sound/soc/codecs/cs35l41.c | 71 +- sound/soc/codecs/cs35l45-i2c.c | 4 +- sound/soc/codecs/cs35l45.c | 6 +- sound/soc/codecs/cs35l45.h | 4 +- sound/soc/codecs/cs4234.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs4270.c | 2 - sound/soc/codecs/cs4271.c | 1 - sound/soc/codecs/cs42l42.c | 4 +- sound/soc/codecs/cs42l51.c | 1 - sound/soc/codecs/cs42l52.c | 1 - sound/soc/codecs/cs42l56.c | 1 - sound/soc/codecs/cs42l73.c | 1 - sound/soc/codecs/cs42xx8.c | 1 - sound/soc/codecs/cs43130.c | 1 - sound/soc/codecs/cs4341.c | 1 - sound/soc/codecs/cs4349.c | 1 - sound/soc/codecs/cs47l15.c | 1 - sound/soc/codecs/cs47l24.c | 1 - sound/soc/codecs/cs47l35.c | 1 - sound/soc/codecs/cs47l85.c | 1 - sound/soc/codecs/cs47l90.c | 1 - sound/soc/codecs/cs47l92.c | 1 - sound/soc/codecs/cs53l30.c | 1 - sound/soc/codecs/cx20442.c | 1 - sound/soc/codecs/cx2072x.c | 17 +- sound/soc/codecs/da7210.c | 3 +- sound/soc/codecs/da7213.c | 1 - sound/soc/codecs/da7218.c | 1 - sound/soc/codecs/da7219.c | 7 - sound/soc/codecs/da732x.c | 7 - sound/soc/codecs/da9055.c | 1 - sound/soc/codecs/dmic.c | 1 - sound/soc/codecs/es7134.c | 1 - sound/soc/codecs/es7241.c | 1 - sound/soc/codecs/es8316.c | 21 +- sound/soc/codecs/es8328.c | 1 - sound/soc/codecs/gtm601.c | 1 - sound/soc/codecs/hda-dai.c | 102 + sound/soc/codecs/hda.c | 395 +++ sound/soc/codecs/hda.h | 19 + sound/soc/codecs/hdac_hdmi.c | 1 - sound/soc/codecs/hdmi-codec.c | 19 +- sound/soc/codecs/ics43432.c | 1 - sound/soc/codecs/inno_rk3036.c | 1 - sound/soc/codecs/isabelle.c | 1 - sound/soc/codecs/jz4740.c | 2 - sound/soc/codecs/lm49453.c | 7 - sound/soc/codecs/lochnagar-sc.c | 1 - sound/soc/codecs/max98088.c | 33 +- sound/soc/codecs/max98090.c | 13 +- sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/max98357a.c | 1 - sound/soc/codecs/max98371.c | 1 - sound/soc/codecs/max98373.c | 16 +- sound/soc/codecs/max98390.c | 3 +- sound/soc/codecs/max98396.c | 271 +- sound/soc/codecs/max98396.h | 10 +- sound/soc/codecs/max9850.c | 1 - sound/soc/codecs/max98520.c | 1 - sound/soc/codecs/max9860.c | 7 +- sound/soc/codecs/max9867.c | 1 - sound/soc/codecs/max98925.c | 1 - sound/soc/codecs/max98926.c | 1 - sound/soc/codecs/max98927.c | 1 - sound/soc/codecs/mc13783.c | 1 - sound/soc/codecs/ml26124.c | 1 - sound/soc/codecs/msm8916-wcd-analog.c | 1 - sound/soc/codecs/msm8916-wcd-digital.c | 47 +- sound/soc/codecs/mt6358.c | 1 + sound/soc/codecs/mt6359-accdet.c | 1 + sound/soc/codecs/mt6359.c | 1 + sound/soc/codecs/nau8315.c | 1 - sound/soc/codecs/nau8540.c | 1 - sound/soc/codecs/nau8810.c | 1 - sound/soc/codecs/nau8821.c | 36 +- sound/soc/codecs/nau8821.h | 1 + sound/soc/codecs/nau8822.c | 15 +- sound/soc/codecs/nau8822.h | 2 + sound/soc/codecs/nau8824.c | 1 - sound/soc/codecs/nau8825.c | 3 +- sound/soc/codecs/pcm1681.c | 1 - sound/soc/codecs/pcm1789.c | 1 - sound/soc/codecs/pcm179x.c | 1 - sound/soc/codecs/pcm186x.c | 2 - sound/soc/codecs/pcm3008.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/pcm5102a.c | 1 - sound/soc/codecs/pcm512x.c | 1 - sound/soc/codecs/rk3328_codec.c | 6 +- sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/rt1011.c | 1 - sound/soc/codecs/rt1015.c | 1 - sound/soc/codecs/rt1015p.c | 1 - sound/soc/codecs/rt1016.c | 1 - sound/soc/codecs/rt1019.c | 1 - sound/soc/codecs/rt1305.c | 1 - sound/soc/codecs/rt1308-sdw.c | 12 + sound/soc/codecs/rt1308.c | 1 - sound/soc/codecs/rt1316-sdw.c | 12 + sound/soc/codecs/rt274.c | 11 +- sound/soc/codecs/rt286.c | 19 +- sound/soc/codecs/rt286.h | 2 - sound/soc/codecs/rt298.c | 61 +- sound/soc/codecs/rt298.h | 2 - sound/soc/codecs/rt5514.c | 1 - sound/soc/codecs/rt5616.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/rt5640.c | 13 +- sound/soc/codecs/rt5645.c | 1 - sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5659.c | 1 - sound/soc/codecs/rt5660.c | 1 - sound/soc/codecs/rt5663.c | 1 - sound/soc/codecs/rt5665.c | 1 - sound/soc/codecs/rt5668.c | 1 - sound/soc/codecs/rt5670.c | 1 - sound/soc/codecs/rt5677.c | 1 - sound/soc/codecs/rt5682.c | 1 - sound/soc/codecs/rt5682s.c | 1 - sound/soc/codecs/rt700.c | 5 + sound/soc/codecs/rt711-sdca.c | 5 + sound/soc/codecs/rt711.c | 5 + sound/soc/codecs/rt715-sdca.c | 12 + sound/soc/codecs/rt715.c | 12 + sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/si476x.c | 1 - sound/soc/codecs/spdif_receiver.c | 1 - sound/soc/codecs/spdif_transmitter.c | 1 - sound/soc/codecs/ssm2518.c | 5 +- sound/soc/codecs/ssm2602.c | 7 +- sound/soc/codecs/ssm4567.c | 5 +- sound/soc/codecs/sta32x.c | 5 +- sound/soc/codecs/sta350.c | 5 +- sound/soc/codecs/sta529.c | 1 - sound/soc/codecs/stac9766.c | 2 - sound/soc/codecs/sti-sas.c | 7 +- sound/soc/codecs/tas2552.c | 13 +- sound/soc/codecs/tas2562.c | 2 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 7 +- sound/soc/codecs/tas2780.c | 663 +++++ sound/soc/codecs/tas2780.h | 101 + sound/soc/codecs/tas5086.c | 3 +- sound/soc/codecs/tas571x.c | 1 - sound/soc/codecs/tas5720.c | 6 +- sound/soc/codecs/tas5805m.c | 1 - sound/soc/codecs/tas6424.c | 7 +- sound/soc/codecs/tfa9879.c | 5 +- sound/soc/codecs/tfa989x.c | 31 +- sound/soc/codecs/tlv320adc3xxx.c | 3 +- sound/soc/codecs/tlv320adcx140.c | 11 +- sound/soc/codecs/tlv320aic23.c | 8 +- sound/soc/codecs/tlv320aic26.c | 16 +- sound/soc/codecs/tlv320aic31xx.c | 20 +- sound/soc/codecs/tlv320aic32x4.c | 11 +- sound/soc/codecs/tlv320aic3x.c | 12 +- sound/soc/codecs/tlv320dac33.c | 13 +- sound/soc/codecs/tscs42xx.c | 1 - sound/soc/codecs/twl4030.c | 102 +- sound/soc/codecs/twl6040.c | 1 - sound/soc/codecs/uda1334.c | 3 +- sound/soc/codecs/uda134x.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wcd-mbhc-v2.c | 12 +- sound/soc/codecs/wcd9335.c | 98 +- sound/soc/codecs/wl1273.c | 1 - sound/soc/codecs/wm0010.c | 1 - sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm5102.c | 1 - sound/soc/codecs/wm5110.c | 1 - sound/soc/codecs/wm8350.c | 1 - sound/soc/codecs/wm8400.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8524.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8727.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8770.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8782.c | 1 - sound/soc/codecs/wm8804.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 1 - sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm8993.c | 1 - sound/soc/codecs/wm8994.c | 1 - sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 2 - sound/soc/codecs/wm8997.c | 1 - sound/soc/codecs/wm8998.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 1 - sound/soc/codecs/wm9705.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - sound/soc/codecs/wm_adsp.c | 25 +- sound/soc/codecs/wsa881x.c | 16 +- sound/soc/codecs/wsa883x.c | 1511 ++++++++++ sound/soc/codecs/zl38060.c | 1 - sound/soc/dwc/dwc-i2s.c | 15 +- sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/fsl_asrc_dma.c | 19 +- sound/soc/fsl/fsl_aud2htx.c | 3 +- sound/soc/fsl/fsl_audmix.c | 6 +- sound/soc/fsl/fsl_easrc.c | 7 +- sound/soc/fsl/fsl_esai.c | 11 +- sound/soc/fsl/fsl_micfil.c | 55 +- sound/soc/fsl/fsl_micfil.h | 9 + sound/soc/fsl/fsl_mqs.c | 136 +- sound/soc/fsl/fsl_rpmsg.c | 3 +- sound/soc/fsl/fsl_sai.c | 375 ++- sound/soc/fsl/fsl_sai.h | 28 +- sound/soc/fsl/fsl_spdif.c | 51 +- sound/soc/fsl/fsl_ssi.c | 23 +- sound/soc/fsl/fsl_utils.c | 69 + sound/soc/fsl/fsl_utils.h | 7 + sound/soc/fsl/fsl_xcvr.c | 12 +- sound/soc/fsl/imx-audmix.c | 4 +- sound/soc/fsl/imx-audmux.c | 24 +- sound/soc/fsl/imx-card.c | 16 +- sound/soc/fsl/mpc5200_psc_i2s.c | 3 +- sound/soc/fsl/pcm030-audio-fabric.c | 8 +- .../generic/audio-graph-card2-custom-sample.dtsi | 101 +- sound/soc/generic/audio-graph-card2.c | 74 +- sound/soc/generic/simple-card-utils.c | 44 +- sound/soc/generic/test-component.c | 20 +- sound/soc/hisilicon/hi6210-i2s.c | 19 +- sound/soc/img/img-i2s-in.c | 7 +- sound/soc/img/img-i2s-out.c | 21 +- sound/soc/img/img-parallel-out.c | 3 +- sound/soc/img/img-spdif-in.c | 3 +- sound/soc/img/img-spdif-out.c | 3 +- sound/soc/img/pistachio-internal-dac.c | 1 - sound/soc/intel/Kconfig | 5 +- sound/soc/intel/atom/sst-atom-controls.c | 8 +- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/atom/sst/sst_ipc.c | 8 +- sound/soc/intel/avs/Makefile | 3 + sound/soc/intel/avs/boards/Kconfig | 121 + sound/soc/intel/avs/boards/Makefile | 27 + sound/soc/intel/avs/boards/da7219.c | 282 ++ sound/soc/intel/avs/boards/dmic.c | 93 + sound/soc/intel/avs/boards/hdaudio.c | 294 ++ sound/soc/intel/avs/boards/i2s_test.c | 180 ++ sound/soc/intel/avs/boards/max98357a.c | 154 ++ sound/soc/intel/avs/boards/max98373.c | 239 ++ sound/soc/intel/avs/boards/nau8825.c | 353 +++ sound/soc/intel/avs/boards/rt274.c | 310 +++ sound/soc/intel/avs/boards/rt286.c | 281 ++ sound/soc/intel/avs/boards/rt298.c | 281 ++ sound/soc/intel/avs/boards/rt5682.c | 340 +++ sound/soc/intel/avs/boards/ssm4567.c | 271 ++ sound/soc/intel/avs/cldma.c | 12 +- sound/soc/intel/avs/core.c | 13 +- sound/soc/intel/avs/dsp.c | 11 +- sound/soc/intel/avs/ipc.c | 1 + sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 18 +- sound/soc/intel/avs/pcm.c | 2 - sound/soc/intel/avs/topology.c | 27 + sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bdw_rt286.c | 279 ++ sound/soc/intel/boards/broadwell.c | 336 --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcht_nocodec.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 8 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 23 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 21 +- sound/soc/intel/boards/haswell.c | 202 -- sound/soc/intel/boards/hda_dsp_common.c | 4 +- sound/soc/intel/boards/hsw_rt5640.c | 176 ++ sound/soc/intel/boards/kbl_da7219_max98357a.c | 21 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 21 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 21 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 21 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 19 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_cs42l42.c | 109 +- sound/soc/intel/boards/sof_da7219_max98373.c | 23 +- sound/soc/intel/boards/sof_nau8825.c | 23 +- sound/soc/intel/boards/sof_pcm512x.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 24 +- sound/soc/intel/boards/sof_realtek_common.h | 6 +- sound/soc/intel/boards/sof_rt5682.c | 25 +- sound/soc/intel/boards/sof_sdw.c | 34 +- sound/soc/intel/catpt/device.c | 5 +- sound/soc/intel/catpt/pcm.c | 26 +- sound/soc/intel/catpt/sysfs.c | 4 +- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-adl-match.c | 7 + .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 6 +- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 72 + sound/soc/intel/keembay/kmb_platform.c | 18 +- sound/soc/intel/skylake/skl-pcm.c | 5 +- sound/soc/intel/skylake/skl-topology.c | 6 - sound/soc/jz4740/Kconfig | 2 +- sound/soc/jz4740/jz4740-i2s.c | 43 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 9 +- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 651 +++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 106 + sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 243 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 150 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 + sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 + sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 871 ++++++ sound/soc/mediatek/mt8186/mt8186-dai-hostless.c | 298 ++ sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 236 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1286 +++++++++ sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 423 +++ sound/soc/mediatek/mt8186/mt8186-dai-src.c | 695 +++++ sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 698 +++++ sound/soc/mediatek/mt8186/mt8186-interconnection.h | 69 + sound/soc/mediatek/mt8186/mt8186-misc-control.c | 252 ++ sound/soc/mediatek/mt8186/mt8186-reg.h | 2913 ++++++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 8 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 6 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 6 +- sound/soc/meson/aiu-acodec-ctrl.c | 1 - sound/soc/meson/aiu-codec-ctrl.c | 1 - sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/axg-frddr.c | 3 + sound/soc/meson/axg-pdm.c | 4 +- sound/soc/meson/axg-spdifin.c | 1 + sound/soc/meson/axg-spdifout.c | 1 + sound/soc/meson/axg-tdm-interface.c | 14 +- sound/soc/meson/axg-toddr.c | 3 + sound/soc/meson/g12a-toacodec.c | 2 - sound/soc/meson/g12a-tohdmitx.c | 1 - sound/soc/meson/meson-codec-glue.c | 2 +- sound/soc/meson/t9015.c | 1 - sound/soc/mxs/mxs-saif.c | 7 +- sound/soc/pxa/magician.c | 8 +- sound/soc/pxa/mmp-sspa.c | 15 +- sound/soc/pxa/pxa-ssp.c | 43 +- sound/soc/pxa/pxa2xx-i2s.c | 27 +- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 1 + sound/soc/qcom/lpass-cpu.c | 7 + sound/soc/qcom/qdsp6/audioreach.c | 4 +- sound/soc/qcom/qdsp6/q6adm.c | 6 +- sound/soc/qcom/qdsp6/q6afe.c | 6 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 23 +- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sc7280.c | 33 + sound/soc/qcom/sdm845.c | 6 +- sound/soc/qcom/sm8250.c | 4 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rockchip_i2s.c | 98 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 13 +- sound/soc/rockchip/rockchip_pdm.c | 7 +- sound/soc/rockchip/rockchip_spdif.c | 1 + sound/soc/samsung/Kconfig | 20 +- sound/soc/samsung/aries_wm8994.c | 7 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 8 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/pcm.c | 7 +- sound/soc/samsung/rx1950_uda1380.c | 4 +- sound/soc/samsung/s3c-i2s-v2.c | 17 +- sound/soc/samsung/s3c2412-i2s.c | 7 +- sound/soc/samsung/s3c24xx-i2s.c | 14 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 7 +- sound/soc/sh/fsi.c | 6 +- sound/soc/sh/hac.c | 3 +- sound/soc/sh/rcar/core.c | 30 +- sound/soc/sh/rcar/ssiu.c | 2 + sound/soc/sh/rz-ssi.c | 11 +- sound/soc/sh/siu_pcm.c | 17 +- sound/soc/sh/ssi.c | 13 +- sound/soc/soc-card.c | 6 + sound/soc/soc-core.c | 167 +- sound/soc/soc-dai.c | 3 +- sound/soc/soc-dapm.c | 10 +- sound/soc/soc-ops.c | 51 +- sound/soc/soc-pcm.c | 3 +- sound/soc/soc-topology-test.c | 37 +- sound/soc/soc-utils.c | 1 - sound/soc/sof/Kconfig | 7 + sound/soc/sof/Makefile | 16 +- sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-dsp-offset.h | 2 + sound/soc/sof/amd/acp.c | 36 + sound/soc/sof/amd/acp.h | 4 + sound/soc/sof/amd/pci-rn.c | 4 + sound/soc/sof/amd/renoir.c | 4 + sound/soc/sof/compress.c | 73 +- sound/soc/sof/core.c | 7 +- sound/soc/sof/debug.c | 5 +- sound/soc/sof/imx/Kconfig | 1 + sound/soc/sof/intel/Kconfig | 27 + sound/soc/sof/intel/Makefile | 4 +- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/bdw.c | 7 +- sound/soc/sof/intel/byt.c | 5 +- sound/soc/sof/intel/cnl.c | 39 +- sound/soc/sof/intel/hda-dai.c | 226 +- sound/soc/sof/intel/hda-dsp.c | 26 +- sound/soc/sof/intel/hda-ipc.c | 39 +- sound/soc/sof/intel/hda-loader.c | 28 +- sound/soc/sof/intel/hda-stream.c | 6 +- sound/soc/sof/intel/hda.c | 149 +- sound/soc/sof/intel/hda.h | 70 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/mtl.c | 805 ++++++ sound/soc/sof/intel/mtl.h | 76 + sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-mtl.c | 71 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/intel/shim.h | 2 + sound/soc/sof/intel/tgl.c | 34 +- sound/soc/sof/ipc.c | 24 +- sound/soc/sof/ipc3-dtrace.c | 55 +- sound/soc/sof/ipc3-loader.c | 11 +- sound/soc/sof/ipc3-pcm.c | 8 +- sound/soc/sof/ipc3-topology.c | 73 +- sound/soc/sof/ipc3.c | 36 +- sound/soc/sof/ipc4-control.c | 216 ++ sound/soc/sof/ipc4-pcm.c | 230 ++ sound/soc/sof/ipc4-priv.h | 7 + sound/soc/sof/ipc4-topology.c | 1811 ++++++++++++ sound/soc/sof/ipc4-topology.h | 245 ++ sound/soc/sof/ipc4.c | 45 +- sound/soc/sof/mediatek/Kconfig | 1 + sound/soc/sof/mediatek/adsp_helper.h | 1 + sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 4 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 7 + sound/soc/sof/mediatek/mt8195/mt8195.c | 50 +- sound/soc/sof/mediatek/mt8195/mt8195.h | 5 + sound/soc/sof/ops.h | 6 + sound/soc/sof/pcm.c | 13 + sound/soc/sof/sof-audio.h | 12 + sound/soc/sof/sof-client-ipc-msg-injector.c | 29 +- sound/soc/sof/sof-client-probes.c | 5 + sound/soc/sof/sof-client.c | 4 +- sound/soc/sof/sof-priv.h | 14 +- sound/soc/sof/topology.c | 121 +- sound/soc/spear/spdif_in.c | 3 +- sound/soc/spear/spdif_out.c | 3 +- sound/soc/sti/sti_uniperif.c | 3 +- sound/soc/stm/stm32_adfsdm.c | 3 +- sound/soc/stm/stm32_i2s.c | 9 +- sound/soc/stm/stm32_sai_sub.c | 11 +- sound/soc/stm/stm32_spdifrx.c | 1 + sound/soc/sunxi/sun4i-codec.c | 85 +- sound/soc/sunxi/sun4i-i2s.c | 23 +- sound/soc/sunxi/sun4i-spdif.c | 3 +- sound/soc/sunxi/sun50i-codec-analog.c | 8 + sound/soc/sunxi/sun8i-codec.c | 7 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra20_ac97.c | 5 +- sound/soc/tegra/tegra20_das.c | 198 +- sound/soc/tegra/tegra20_das.h | 120 - sound/soc/tegra/tegra20_i2s.c | 9 +- sound/soc/tegra/tegra20_spdif.c | 1 + sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_i2s.c | 7 +- sound/soc/tegra/tegra210_mbdrc.c | 1014 +++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 ++ sound/soc/tegra/tegra210_ope.c | 419 +++ sound/soc/tegra/tegra210_ope.h | 90 + sound/soc/tegra/tegra210_peq.c | 434 +++ sound/soc/tegra/tegra210_peq.h | 56 + sound/soc/tegra/tegra30_i2s.c | 9 +- sound/soc/ti/davinci-i2s.c | 35 +- sound/soc/ti/davinci-mcasp.c | 16 +- sound/soc/ti/davinci-vcif.c | 3 +- sound/soc/ti/omap-dmic.c | 3 +- sound/soc/ti/omap-hdmi.c | 1 + sound/soc/ti/omap-mcbsp.c | 15 +- sound/soc/ti/omap-mcpdm.c | 7 +- sound/soc/uniphier/evea.c | 1 - sound/soc/ux500/mop500.c | 2 - sound/soc/ux500/mop500_ab8500.c | 2 - sound/soc/ux500/mop500_ab8500.h | 2 - sound/soc/ux500/ux500_msp_dai.c | 41 +- sound/soc/ux500/ux500_msp_dai.h | 2 - sound/soc/ux500/ux500_msp_i2s.c | 2 - sound/soc/ux500/ux500_msp_i2s.h | 2 - sound/soc/ux500/ux500_pcm.c | 2 - sound/soc/ux500/ux500_pcm.h | 2 - sound/soc/xilinx/xlnx_formatter_pcm.c | 18 +- sound/soc/xilinx/xlnx_i2s.c | 1 + sound/soc/xilinx/xlnx_spdif.c | 1 + sound/soc/xtensa/xtfpga-i2s.c | 19 +- 679 files changed, 27894 insertions(+), 3270 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml delete mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml create mode 100644 Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/tas2780.yaml create mode 100644 include/dt-bindings/sound/qcom,wcd9335.h create mode 100644 sound/soc/amd/acp-es8336.c create mode 100644 sound/soc/amd/acp/acp-rembrandt.c create mode 100644 sound/soc/codecs/hda-dai.c create mode 100644 sound/soc/codecs/hda.c create mode 100644 sound/soc/codecs/hda.h create mode 100644 sound/soc/codecs/tas2780.c create mode 100644 sound/soc/codecs/tas2780.h create mode 100644 sound/soc/codecs/wsa883x.c create mode 100644 sound/soc/intel/avs/boards/Kconfig create mode 100644 sound/soc/intel/avs/boards/Makefile create mode 100644 sound/soc/intel/avs/boards/da7219.c create mode 100644 sound/soc/intel/avs/boards/dmic.c create mode 100644 sound/soc/intel/avs/boards/hdaudio.c create mode 100644 sound/soc/intel/avs/boards/i2s_test.c create mode 100644 sound/soc/intel/avs/boards/max98357a.c create mode 100644 sound/soc/intel/avs/boards/max98373.c create mode 100644 sound/soc/intel/avs/boards/nau8825.c create mode 100644 sound/soc/intel/avs/boards/rt274.c create mode 100644 sound/soc/intel/avs/boards/rt286.c create mode 100644 sound/soc/intel/avs/boards/rt298.c create mode 100644 sound/soc/intel/avs/boards/rt5682.c create mode 100644 sound/soc/intel/avs/boards/ssm4567.c create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-mtl-match.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h create mode 100644 sound/soc/sof/intel/mtl.c create mode 100644 sound/soc/sof/intel/mtl.h create mode 100644 sound/soc/sof/intel/pci-mtl.c create mode 100644 sound/soc/sof/ipc4-control.c create mode 100644 sound/soc/sof/ipc4-pcm.c create mode 100644 sound/soc/sof/ipc4-topology.c create mode 100644 sound/soc/sof/ipc4-topology.h delete mode 100644 sound/soc/tegra/tegra20_das.h create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h

2 1

[PATCH] ASoC: atmel_ssc_dai: remove #ifdef CONFIG_PM
by Claudiu Beznea 15 Jul '22

15 Jul '22

Remove #ifdef CONFIG_PM and use pm_ptr() instead. Signed-off-by: Claudiu Beznea <claudiu.beznea(a)microchip.com> --- sound/soc/atmel/atmel_ssc_dai.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 8aae0beadcfe..e868b7e028d6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -762,7 +762,6 @@ static int atmel_ssc_trigger(struct snd_pcm_substream *substream, return 0; } -#ifdef CONFIG_PM static int atmel_ssc_suspend(struct snd_soc_component *component) { struct atmel_ssc_info *ssc_p; @@ -821,10 +820,6 @@ static int atmel_ssc_resume(struct snd_soc_component *component) return 0; } -#else /* CONFIG_PM */ -# define atmel_ssc_suspend NULL -# define atmel_ssc_resume NULL -#endif /* CONFIG_PM */ #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -859,8 +854,8 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { static const struct snd_soc_component_driver atmel_ssc_component = { .name = "atmel-ssc", - .suspend = atmel_ssc_suspend, - .resume = atmel_ssc_resume, + .suspend = pm_ptr(atmel_ssc_suspend), + .resume = pm_ptr(atmel_ssc_resume), .legacy_dai_naming = 1, }; -- 2.34.1

2 1

[GIT PULL] ASoC fixes for v5.19-rc4-2
by Mark Brown 15 Jul '22

15 Jul '22

The following changes since commit 980555e95f7cabdc9c80a07107622b097ba23703: ASoC: madera: Fix event generation for rate controls (2022-06-24 16:22:01 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.19-rc4-2 for you to fetch changes up to 1e347f861da8ddb17e1d1b3113cb6c188e0de3e5: ASoC: rockchip-i2s: Undo BCLK pinctrl changes (2022-07-14 13:25:52 +0100) ---------------------------------------------------------------- ASoC: Drop Rockchip BCLK management for v5.19 As covered in the second revert commit in this pull request the version of the BCLK muxing that's in v5.19 is causing issues, let's just revert it and wait for the more complete support in v5.20 instead. ---------------------------------------------------------------- Alexandru Elisei (1): ASoC: rockchip: i2s: Fix NULL pointer dereference when pinctrl is not found Brent Lu (1): ASoC: Intel: sof_rt5682: fix out-of-bounds array access Bryan O'Donoghue (1): ASoC: dt-bindings: Fix description for msm8916 Charles Keepax (4): ASoC: wm5102: Fix event generation for output compensation ASoC: wm8998: Fix event generation for input mux ASoC: cs47l92: Fix event generation for OUT1 demux ASoC: arizona: Update arizona_aif_cfg_changed to use RX_BCLK_RATE Daniel Mack (1): ASoC: max98396: Fix register access for PCM format settings David Owens (1): ASoC: ti: omap-mcbsp: duplicate sysfs error Francesco Dolcini (1): ASoC: sgtl5000: Fix noise on shutdown/remove Hector Martin (2): ASoC: tas2764: Correct playback volume range ASoC: tas2764: Fix amp gain register offset & default Jianglei Nie (1): ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph() Marek Vasut (1): ASoC: doc: Capitalize RESET line name Mark Brown (1): ASoC: rockchip-i2s: Undo BCLK pinctrl changes Martin Povišer (2): ASoC: tas2764: Add post reset delays ASoC: tas2764: Fix and extend FSYNC polarity handling Oder Chiou (1): ASoC: rt5640: Fix the wrong state of JD1 and JD2 Peter Ujfalusi (2): ASoC: Intel: Skylake: Correct the ssp rate discovery in skl_get_ssp_clks() ASoC: Intel: Skylake: Correct the handling of fmt_config flexible array Sascha Hauer (2): ASoC: audio_graph_card2: Fix port numbers in example ASoC: tlv320adcx140: Fix tx_mask check Yassine Oudjana (1): ASoC: wcd9335: Use int array instead of bitmask for TX mixers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 8 +- Documentation/sound/soc/dai.rst | 2 +- sound/soc/codecs/arizona.c | 4 +- sound/soc/codecs/cs47l92.c | 8 +- sound/soc/codecs/max98396.c | 10 +- sound/soc/codecs/rt5640.c | 30 ++-- sound/soc/codecs/sgtl5000.c | 9 ++ sound/soc/codecs/sgtl5000.h | 1 + sound/soc/codecs/tas2764.c | 46 +++--- sound/soc/codecs/tas2764.h | 6 +- sound/soc/codecs/tlv320adcx140.c | 13 +- sound/soc/codecs/wcd9335.c | 17 ++- sound/soc/codecs/wm5102.c | 21 ++- sound/soc/codecs/wm8998.c | 21 ++- sound/soc/generic/audio-graph-card2.c | 6 +- sound/soc/intel/boards/sof_rt5682.c | 10 +- sound/soc/intel/skylake/skl-nhlt.c | 40 ++++-- sound/soc/qcom/qdsp6/q6apm.c | 1 + sound/soc/rockchip/rockchip_i2s.c | 160 ++++----------------- sound/soc/ti/omap-mcbsp-priv.h | 2 - sound/soc/ti/omap-mcbsp-st.c | 14 +- sound/soc/ti/omap-mcbsp.c | 19 +-- 22 files changed, 199 insertions(+), 249 deletions(-)

2 5

stream == SND_PCM_STREAM_PLAYBACK ? _("underrun") : _("overrun"),
by sujith kumar reddy 15 Jul '22

15 Jul '22

Hi ALL, in the below link https://github.com/bear24rw/alsa-utils/blob/master/aplay/aplay.c#L1360 My question here is .. aplay -->writes the data from userspace to system dma (in the kernel alsabuffer).and from system dma to hardware dma(if hardware dma present)and to fifo. Now the application writes the data to system dma -->is always fast because the cpu runs in GHz. and from system dma to i2s fifo runs on i2s mclk.which is running on 196.6MHz. My point is.. cpu is always fast and controller is slow ,so when playback we should get overrun. the command should be like this in my point of you. stream == SND_PCM_STREAM_PLAYBACK ? : _("overrun") _("underrun"), Please let me know if I am wrong. Thanks Sujith

1 0

[PATCH] ASoC: mediatek: mt8195: extract SOF common code
by Chunxu Li 15 Jul '22

15 Jul '22

The functions related to SOF can be reused in different machine drivers, such as mt8195 or mt8186, so extract the common code to avoid duplication. Set mtk_soc_card_data which include machine private data and SOF private data as card drvdata, then the difference between machine private can be ignored such as mt8195_mt6359_priv or mt8186_mt6366_priv, at the same time the SOF related code can be reused in different machine drivers. Signed-off-by: Chunxu Li <chunxu.li(a)mediatek.com> --- sound/soc/mediatek/common/Makefile | 2 +- .../soc/mediatek/common/mtk-dsp-sof-common.c | 196 +++++++++++++++ .../soc/mediatek/common/mtk-dsp-sof-common.h | 35 +++ sound/soc/mediatek/common/mtk-soc-card.h | 17 ++ sound/soc/mediatek/mt8195/mt8195-mt6359.c | 233 ++++-------------- 5 files changed, 300 insertions(+), 183 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dsp-sof-common.c create mode 100644 sound/soc/mediatek/common/mtk-dsp-sof-common.h create mode 100644 sound/soc/mediatek/common/mtk-soc-card.h diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile index acbe01e9e928..576deb7f8cce 100644 --- a/sound/soc/mediatek/common/Makefile +++ b/sound/soc/mediatek/common/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o +snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o obj-$(CONFIG_SND_SOC_MTK_BTCVSD) += mtk-btcvsd.o diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c new file mode 100644 index 000000000000..1098694f4f36 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c @@ -0,0 +1,196 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * mtk-dsp-sof-common.c -- MediaTek dsp sof common ctrl + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Chunxu Li <chunxu.li(a)mediatek.com> + */ + +#include <mtk-dsp-sof-common.h> +#include <mtk-soc-card.h> + +/* fixup the BE DAI link to match any values from topology */ +int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_card *card = rtd->card; + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); + struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + int i, j, ret = 0; + + for (i = 0; i < sof_priv->num_streams; i++) { + struct snd_soc_dai *cpu_dai; + struct snd_soc_pcm_runtime *runtime; + struct snd_soc_dai_link *sof_dai_link = NULL; + const struct sof_conn_stream *conn = &sof_priv->conn_streams[i]; + + if (strcmp(rtd->dai_link->name, conn->normal_link)) + continue; + + for_each_card_rtds(card, runtime) { + if (strcmp(runtime->dai_link->name, conn->sof_link)) + continue; + + for_each_rtd_cpu_dais(runtime, j, cpu_dai) { + if (cpu_dai->stream_active[conn->stream_dir] > 0) { + sof_dai_link = runtime->dai_link; + break; + } + } + break; + } + + if (sof_dai_link && sof_dai_link->be_hw_params_fixup) + ret = sof_dai_link->be_hw_params_fixup(runtime, params); + + break; + } + + return ret; +} +EXPORT_SYMBOL_GPL(mtk_sof_dai_link_fixup); + +int mtk_sof_card_probe(struct snd_soc_card *card) +{ + int i; + struct snd_soc_dai_link *dai_link; + + /* Set stream_name to help sof bind widgets */ + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->no_pcm && !dai_link->stream_name && dai_link->name) + dai_link->stream_name = dai_link->name; + } + + return 0; +} +EXPORT_SYMBOL_GPL(mtk_sof_card_probe); + +int mtk_sof_card_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_component *sof_comp = NULL; + struct mtk_soc_card_data *soc_card_data = + snd_soc_card_get_drvdata(card); + struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + int i; + + /* 1. find sof component */ + for_each_card_rtds(card, rtd) { + sof_comp = snd_soc_rtdcom_lookup(rtd, "sof-audio-component"); + if (sof_comp) + break; + } + + if (!sof_comp) { + dev_info(card->dev, "probe without sof-audio-component\n"); + return 0; + } + + /* 2. add route path and fixup callback */ + for (i = 0; i < sof_priv->num_streams; i++) { + const struct sof_conn_stream *conn = &sof_priv->conn_streams[i]; + struct snd_soc_pcm_runtime *sof_rtd = NULL; + struct snd_soc_pcm_runtime *normal_rtd = NULL; + + for_each_card_rtds(card, rtd) { + if (!strcmp(rtd->dai_link->name, conn->sof_link)) { + sof_rtd = rtd; + continue; + } + if (!strcmp(rtd->dai_link->name, conn->normal_link)) { + normal_rtd = rtd; + continue; + } + if (normal_rtd && sof_rtd) + break; + } + if (normal_rtd && sof_rtd) { + int j; + struct snd_soc_dai *cpu_dai; + + for_each_rtd_cpu_dais(sof_rtd, j, cpu_dai) { + struct snd_soc_dapm_route route; + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_widget *play_widget = + cpu_dai->playback_widget; + struct snd_soc_dapm_widget *cap_widget = + cpu_dai->capture_widget; + memset(&route, 0, sizeof(route)); + if (conn->stream_dir == SNDRV_PCM_STREAM_CAPTURE && + cap_widget) { + snd_soc_dapm_widget_for_each_sink_path(cap_widget, p) { + route.source = conn->sof_dma; + route.sink = p->sink->name; + snd_soc_dapm_add_routes(&card->dapm, &route, 1); + } + } else if (conn->stream_dir == SNDRV_PCM_STREAM_PLAYBACK && + play_widget) { + snd_soc_dapm_widget_for_each_source_path(play_widget, p) { + route.source = p->source->name; + route.sink = conn->sof_dma; + snd_soc_dapm_add_routes(&card->dapm, &route, 1); + } + } else { + dev_err(cpu_dai->dev, "stream dir and widget not pair\n"); + } + } + + sof_rtd->dai_link->be_hw_params_fixup = + sof_comp->driver->be_hw_params_fixup; + if (sof_priv->sof_dai_link_fixup) + normal_rtd->dai_link->be_hw_params_fixup = + sof_priv->sof_dai_link_fixup; + else + normal_rtd->dai_link->be_hw_params_fixup = mtk_sof_dai_link_fixup; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(mtk_sof_card_late_probe); + +int mtk_sof_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, + const char *propname, struct snd_soc_dai_link *pre_dai_links, + int pre_num_links) +{ + struct device *dev = card->dev; + struct snd_soc_dai_link *parsed_dai_link; + const char *dai_name = NULL; + int i, j, ret, num_links, parsed_num_links = 0; + + num_links = of_property_count_strings(np, "mediatek,dai-link"); + if (num_links < 0 || num_links > card->num_links) { + dev_dbg(dev, "number of dai-link is invalid\n"); + return -EINVAL; + } + + parsed_dai_link = devm_kcalloc(dev, num_links, sizeof(*parsed_dai_link), GFP_KERNEL); + if (!parsed_dai_link) + return -ENOMEM; + + for (i = 0; i < num_links; i++) { + ret = of_property_read_string_index(np, propname, i, &dai_name); + if (ret) { + dev_dbg(dev, "ASoC: Property '%s' index %d could not be read: %d\n", + propname, i, ret); + return ret; + } + dev_dbg(dev, "ASoC: Property get dai_name:%s\n", dai_name); + for (j = 0; j < pre_num_links; j++) { + if (!strcmp(dai_name, pre_dai_links[j].name)) { + memcpy(&parsed_dai_link[parsed_num_links++], &pre_dai_links[j], + sizeof(struct snd_soc_dai_link)); + break; + } + } + } + + if (parsed_num_links != num_links) + return -EINVAL; + + card->dai_link = parsed_dai_link; + card->num_links = parsed_num_links; + + return 0; +} +EXPORT_SYMBOL_GPL(mtk_sof_dailink_parse_of); diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.h b/sound/soc/mediatek/common/mtk-dsp-sof-common.h new file mode 100644 index 000000000000..f02ba2ce3718 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.h @@ -0,0 +1,35 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * mtk-dsp-sof-common.h -- MediaTek dsp sof common definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Chunxu Li <chunxu.li(a)mediatek.com> + */ + +#ifndef _MTK_DSP_SOF_COMMON_H_ +#define _MTK_DSP_SOF_COMMON_H_ + +#include <sound/soc.h> +struct sof_conn_stream { + const char *normal_link; + const char *sof_link; + const char *sof_dma; + int stream_dir; +}; + +struct mtk_sof_priv { + const struct sof_conn_stream *conn_streams; + int num_streams; + int (*sof_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); +}; + +int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); +int mtk_sof_card_probe(struct snd_soc_card *card); +int mtk_sof_card_late_probe(struct snd_soc_card *card); +int mtk_sof_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, + const char *propname, struct snd_soc_dai_link *pre_dai_links, + int pre_num_links); + +#endif diff --git a/sound/soc/mediatek/common/mtk-soc-card.h b/sound/soc/mediatek/common/mtk-soc-card.h new file mode 100644 index 000000000000..eeda79370049 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-soc-card.h @@ -0,0 +1,17 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * mtk-soc-card.h -- MediaTek soc card data definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Chunxu Li <chunxu.li(a)mediatek.com> + */ + +#ifndef _MTK_SOC_CARD_H_ +#define _MTK_SOC_CARD_H_ + +struct mtk_soc_card_data { + void *mach_priv; + void *sof_priv; +}; + +#endif diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 54a00b0699b1..c530e3fc27e4 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -20,6 +20,8 @@ #include "../../codecs/rt1011.h" #include "../../codecs/rt5682.h" #include "../common/mtk-afe-platform-driver.h" +#include "../common/mtk-dsp-sof-common.h" +#include "../common/mtk-soc-card.h" #include "mt8195-afe-clk.h" #include "mt8195-afe-common.h" @@ -54,13 +56,6 @@ struct mt8195_card_data { unsigned long quirk; }; -struct sof_conn_stream { - const char *normal_link; - const char *sof_link; - const char *sof_dma; - int stream_dir; -}; - struct mt8195_mt6359_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; @@ -374,7 +369,8 @@ static const struct snd_soc_ops mt8195_dptx_ops = { static int mt8195_dptx_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; int ret; @@ -389,7 +385,8 @@ static int mt8195_dptx_codec_init(struct snd_soc_pcm_runtime *rtd) static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; int ret; @@ -555,7 +552,8 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *cmpnt_codec = asoc_rtd_to_codec(rtd, 0)->component; - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; struct snd_soc_jack *jack = &priv->headset_jack; struct snd_soc_component *cmpnt_afe = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); @@ -722,7 +720,8 @@ static int mt8195_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { struct snd_soc_component *component = dapm->component; - struct mt8195_mt6359_priv *priv = snd_soc_card_get_drvdata(card); + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); + struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; int ret; /* @@ -1321,175 +1320,24 @@ static struct snd_soc_card mt8195_mt6359_soc_card = { static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai_link *sof_dai_link = NULL; - struct snd_soc_pcm_runtime *runtime; - struct snd_soc_dai *cpu_dai; - int i, j, ret = 0; - - for (i = 0; i < ARRAY_SIZE(g_sof_conn_streams); i++) { - const struct sof_conn_stream *conn = &g_sof_conn_streams[i]; - - if (strcmp(rtd->dai_link->name, conn->normal_link)) - continue; - - for_each_card_rtds(card, runtime) { - if (strcmp(runtime->dai_link->name, conn->sof_link)) - continue; - - for_each_rtd_cpu_dais(runtime, j, cpu_dai) { - if (cpu_dai->stream_active[conn->stream_dir] > 0) { - sof_dai_link = runtime->dai_link; - break; - } - } - break; - } + int ret; - if (sof_dai_link && sof_dai_link->be_hw_params_fixup) - ret = sof_dai_link->be_hw_params_fixup(runtime, params); - - break; - } + ret = mtk_sof_dai_link_fixup(rtd, params); if (!strcmp(rtd->dai_link->name, "ETDM2_IN_BE") || !strcmp(rtd->dai_link->name, "ETDM1_OUT_BE")) { - mt8195_etdm_hw_params_fixup(runtime, params); + mt8195_etdm_hw_params_fixup(rtd, params); } return ret; } -static int mt8195_mt6359_card_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *runtime; - struct snd_soc_component *sof_comp = NULL; - int i; - - /* 1. find sof component */ - for_each_card_rtds(card, runtime) { - for (i = 0; i < runtime->num_components; i++) { - if (!runtime->components[i]->driver->name) - continue; - if (!strcmp(runtime->components[i]->driver->name, "sof-audio-component")) { - sof_comp = runtime->components[i]; - break; - } - } - } - - if (!sof_comp) { - dev_info(card->dev, " probe without component\n"); - return 0; - } - /* 2. add route path and fixup callback */ - for (i = 0; i < ARRAY_SIZE(g_sof_conn_streams); i++) { - const struct sof_conn_stream *conn = &g_sof_conn_streams[i]; - struct snd_soc_pcm_runtime *sof_rtd = NULL; - struct snd_soc_pcm_runtime *normal_rtd = NULL; - struct snd_soc_pcm_runtime *rtd = NULL; - - for_each_card_rtds(card, rtd) { - if (!strcmp(rtd->dai_link->name, conn->sof_link)) { - sof_rtd = rtd; - continue; - } - if (!strcmp(rtd->dai_link->name, conn->normal_link)) { - normal_rtd = rtd; - continue; - } - if (normal_rtd && sof_rtd) - break; - } - if (normal_rtd && sof_rtd) { - int j; - struct snd_soc_dai *cpu_dai; - - for_each_rtd_cpu_dais(sof_rtd, j, cpu_dai) { - struct snd_soc_dapm_route route; - struct snd_soc_dapm_path *p = NULL; - struct snd_soc_dapm_widget *play_widget = - cpu_dai->playback_widget; - struct snd_soc_dapm_widget *cap_widget = - cpu_dai->capture_widget; - memset(&route, 0, sizeof(route)); - if (conn->stream_dir == SNDRV_PCM_STREAM_CAPTURE && - cap_widget) { - snd_soc_dapm_widget_for_each_sink_path(cap_widget, p) { - route.source = conn->sof_dma; - route.sink = p->sink->name; - snd_soc_dapm_add_routes(&card->dapm, &route, 1); - } - } else if (conn->stream_dir == SNDRV_PCM_STREAM_PLAYBACK && - play_widget){ - snd_soc_dapm_widget_for_each_source_path(play_widget, p) { - route.source = p->source->name; - route.sink = conn->sof_dma; - snd_soc_dapm_add_routes(&card->dapm, &route, 1); - } - } else { - dev_err(cpu_dai->dev, "stream dir and widget not pair\n"); - } - } - normal_rtd->dai_link->be_hw_params_fixup = mt8195_dai_link_fixup; - } - } - - return 0; -} - -static int mt8195_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, - const char *propname) -{ - struct device *dev = card->dev; - struct snd_soc_dai_link *link; - const char *dai_name = NULL; - int i, j, ret, num_links; - - num_links = of_property_count_strings(np, "mediatek,dai-link"); - - if (num_links < 0 || num_links > ARRAY_SIZE(mt8195_mt6359_dai_links)) { - dev_dbg(dev, "number of dai-link is invalid\n"); - return -EINVAL; - } - - card->dai_link = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL); - if (!card->dai_link) - return -ENOMEM; - - card->num_links = 0; - link = card->dai_link; - - for (i = 0; i < num_links; i++) { - ret = of_property_read_string_index(np, propname, i, &dai_name); - if (ret) { - dev_dbg(dev, "ASoC: Property '%s' index %d could not be read: %d\n", - propname, i, ret); - return -EINVAL; - } - - for (j = 0; j < ARRAY_SIZE(mt8195_mt6359_dai_links); j++) { - if (!strcmp(dai_name, mt8195_mt6359_dai_links[j].name)) { - memcpy(link, &mt8195_mt6359_dai_links[j], - sizeof(struct snd_soc_dai_link)); - link++; - card->num_links++; - break; - } - } - } - - if (card->num_links != num_links) - return -EINVAL; - - return 0; -} - static int mt8195_mt6359_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8195_mt6359_soc_card; struct snd_soc_dai_link *dai_link; - struct mt8195_mt6359_priv *priv; + struct mtk_soc_card_data *soc_card_data; + struct mt8195_mt6359_priv *mach_priv; struct device_node *platform_node, *adsp_node, *dp_node, *hdmi_node; struct mt8195_card_data *card_data; int is5682s = 0; @@ -1512,17 +1360,41 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) if (strstr(card->name, "_5682s")) is5682s = 1; + soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL); + if (!soc_card_data) + return -ENOMEM; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) + mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL); + if (!mach_priv) return -ENOMEM; + soc_card_data->mach_priv = mach_priv; + + adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); + if (adsp_node) { + struct mtk_sof_priv *sof_priv; + + sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); + if (!sof_priv) { + ret = -ENOMEM; + goto err_kzalloc; + } + sof_priv->conn_streams = g_sof_conn_streams; + sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); + sof_priv->sof_dai_link_fixup = mt8195_dai_link_fixup; + soc_card_data->sof_priv = sof_priv; + card->late_probe = mtk_sof_card_late_probe; + sof_on = 1; + } + if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { - ret = mt8195_dailink_parse_of(card, pdev->dev.of_node, - "mediatek,dai-link"); + ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, + "mediatek,dai-link", + mt8195_mt6359_dai_links, + ARRAY_SIZE(mt8195_mt6359_dai_links)); if (ret) { dev_dbg(&pdev->dev, "Parse dai-link fail\n"); - return -EINVAL; + goto err_parse_of; } } else { if (!sof_on) @@ -1533,13 +1405,10 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) "mediatek,platform", 0); if (!platform_node) { dev_dbg(&pdev->dev, "Property 'platform' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto err_platform_node; } - adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); - if (adsp_node) - sof_on = 1; - dp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,dptx-codec", 0); hdmi_node = of_parse_phandle(pdev->dev.of_node, "mediatek,hdmi-codec", 0); @@ -1612,17 +1481,17 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) } } - if (sof_on) - card->late_probe = mt8195_mt6359_card_late_probe; - - snd_soc_card_set_drvdata(card, priv); + snd_soc_card_set_drvdata(card, soc_card_data); ret = devm_snd_soc_register_card(&pdev->dev, card); of_node_put(platform_node); - of_node_put(adsp_node); of_node_put(dp_node); of_node_put(hdmi_node); +err_kzalloc: +err_parse_of: +err_platform_node: + of_node_put(adsp_node); return ret; } -- 2.25.1

2 1

[PATCH] ALSA: bcd2000: Fix a UAF bug on the error path of probing
by Zheyu Ma 15 Jul '22

15 Jul '22

When the driver fails in snd_card_register() at probe time, it will free the 'bcd2k->midi_out_urb' before killing it, which may cause a UAF bug. The following log can reveal it: [ 50.727020] BUG: KASAN: use-after-free in bcd2000_input_complete+0x1f1/0x2e0 [snd_bcd2000] [ 50.727623] Read of size 8 at addr ffff88810fab0e88 by task swapper/4/0 [ 50.729530] Call Trace: [ 50.732899] bcd2000_input_complete+0x1f1/0x2e0 [snd_bcd2000] Fix this by adding usb_kill_urb() before usb_free_urb(). Fixes: b47a22290d58 ("ALSA: MIDI driver for Behringer BCD2000 USB device") Signed-off-by: Zheyu Ma <zheyuma97(a)gmail.com> --- sound/usb/bcd2000/bcd2000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c index cd4a0bc6d278..7aec0a95c609 100644 --- a/sound/usb/bcd2000/bcd2000.c +++ b/sound/usb/bcd2000/bcd2000.c @@ -348,7 +348,8 @@ static int bcd2000_init_midi(struct bcd2000 *bcd2k) static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, struct usb_interface *interface) { - /* usb_kill_urb not necessary, urb is aborted automatically */ + usb_kill_urb(bcd2k->midi_out_urb); + usb_kill_urb(bcd2k->midi_in_urb); usb_free_urb(bcd2k->midi_out_urb); usb_free_urb(bcd2k->midi_in_urb); -- 2.25.1

2 1

[PATCH] ASoC: codecs: add support for ES8326
by Zhu Ning 15 Jul '22

15 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 794 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 +++++++++ 4 files changed, 988 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..d62c596ac878 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,794 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS_1B, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS_1B, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_RATES SNDRV_PCM_RATE_8000_96000 + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DETECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + es8326_disable_micbias(comp); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if(!es8326->jack) + goto out; + es8326_enable_micbias(comp); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_OFF); + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) + snd_soc_jack_report(es8326->jack, 0, SND_JACK_BTN_0); + + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..9affd5c14407 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.c -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

1 0

No Audio: MAG Z690 TOMAHAWK WIFI DDR4 & how to build info request
by GitHub issues - edited 14 Jul '22

14 Jul '22

alsa-project/alsa-ucm-conf issue #189 was edited from Dave--G: My MAG Z690 TOMAHAWK WIFI DDR4 motherboard has no alsa audio support. It looks like the realtek-alc4080 audio chip is supported, just not on this motherboard due to a missing entry in `USB-Audio.conf`. The USB audio device id is `0db0:b202`. Currently it just displays as `USB Audio` and doesn't function at all. This device works under windows so the hardware is functioning but no linux distro I've tried. I attempted to address this myself, by downloading `https://github.com/alsa-project/alsa-ucm-conf/releases/tag/v1.2.7.2`, copying it to `/usr/share/alsa` (preserving simlinks of course) and editing `ucm2/USB-Audio/USB-Audio.conf`, changing line 46 to read ` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|b202)))"`. However, when I execute `sudo alsa force-reload` it generates parsing errors and `pactl list sinks` does not show any changes to the device properties as expected. Can someone please explain what I need to do to get this device working? Am I on the right track? Do I need to rebuild alsa from github? It's not clear and I'm not seeing build instructions. I've been struggling with this for quite some time so any help would be appreciated. I will PR the fix back to this repo once its working. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/189 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Audio: MAG Z690 TOMAHAWK WIFI DDR4 & how to build info request
by GitHub issues - opened 14 Jul '22

14 Jul '22

alsa-project/alsa-ucm-conf issue #189 was opened from Dave--G: My MAG Z690 TOMAHAWK WIFI DDR4 motherboard has no alsa audio support. It looks like the realtek-alc4080 audio chip is supported, just not on this motherboard due to a missing entry in `USB-Audio.conf`. The USB audio device id is `0db0:b202`. Currently it just displays as `USB Audio` and doesn't function at all. This device works under windows so the hardware is functioning but no linux distro I've tried. I attempted to address this myself, by downloading `https://github.com/alsa-project/alsa-ucm-conf/releases/tag/v1.2.7.2`, copying it to `/usr/share/alsa` (preserving simlinks of course) and editing `ucm2/USB-Audio/USB-Audio.conf`, changing line 46 to read ` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|b202)))"`. However, when I execute `sudo alsa force-reload` it generates parsing errors and `pactl list sinks` does not show any changes to the device properties as expected. Can someone please explain what I need to do to get this device working? Am I on the right track? Do I need to rebuild alsa from github? It's not clear and I'm not seeing build instructions. I've been struggling with this for quite some time so any help would be appreciated. I will PR the fix back to this repo once its working. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/189 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl()
by Kuninori Morimoto 14 Jul '22

14 Jul '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Before, ssiu.c didn't care SSI5-8, thus, commit b1384d4c95088d0 ("ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()") cares it for status clear. But we should care it for error irq handling, too. This patch cares it. Reported-by: Nguyen Bao Nguyen <nguyen.nguyen.yj(a)renesas.com> Reported-by: Nishiyama Kunihiko <kunihiko.nishiyama.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..d7f4646ee029 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -67,6 +67,8 @@ static void rsnd_ssiu_busif_err_irq_ctrl(struct rsnd_mod *mod, int enable) shift = 1; offset = 1; break; + default: + return; } for (i = 0; i < 4; i++) { -- 2.25.1

2 1

[PATCH v2] ASoC: SOF: ipc-msg-injector: fix copy in sof_msg_inject_ipc4_dfs_write()
by Dan Carpenter 14 Jul '22

14 Jul '22

There are two bugs that have to do with when we copy the payload: size = simple_write_to_buffer(ipc4_msg->data_ptr, priv->max_msg_size, ppos, buffer, count); The value of "*ppos" was supposed to be zero but it is sizeof(ipc4_msg->header_u64) so it will copy the data into the middle of the "ipc4_msg->data_ptr" buffer instead of to the start. The second problem is "buffer" should be "buffer + sizeof(ipc4_msg->header_u64)". This function is used for fuzz testing so the data is normally random and this bug likely does not affect anyone very much. In this context, it's simpler and more appropriate to use copy_from_user() instead of simple_write_to_buffer() so I have re-written the function. Fixes: 066c67624d8c ("ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- v2: In v1 I missed the "buffer + sizeof(ipc4_msg->header_u64)" bug. sound/soc/sof/sof-client-ipc-msg-injector.c | 29 +++++++++------------ 1 file changed, 12 insertions(+), 17 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index 6bdfa527b7f7..752d5320680f 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -181,7 +181,7 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; struct sof_ipc4_msg *ipc4_msg = priv->tx_buffer; - ssize_t size; + size_t data_size; int ret; if (*ppos) @@ -191,25 +191,20 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, return -EINVAL; /* copy the header first */ - size = simple_write_to_buffer(&ipc4_msg->header_u64, - sizeof(ipc4_msg->header_u64), - ppos, buffer, count); - if (size < 0) - return size; - if (size != sizeof(ipc4_msg->header_u64)) + if (copy_from_user(&ipc4_msg->header_u64, buffer, + sizeof(ipc4_msg->header_u64))) return -EFAULT; - count -= size; + data_size = count - sizeof(ipc4_msg->header_u64); + if (data_size > priv->max_msg_size) + return -EINVAL; + /* Copy the payload */ - size = simple_write_to_buffer(ipc4_msg->data_ptr, - priv->max_msg_size, ppos, buffer, - count); - if (size < 0) - return size; - if (size != count) + if (copy_from_user(ipc4_msg->data_ptr, + buffer + sizeof(ipc4_msg->header_u64), data_size)) return -EFAULT; - ipc4_msg->data_size = count; + ipc4_msg->data_size = data_size; /* Initialize the reply storage */ ipc4_msg = priv->rx_buffer; @@ -221,9 +216,9 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, /* return the error code if test failed */ if (ret < 0) - size = ret; + return ret; - return size; + return count; }; static int sof_msg_inject_dfs_release(struct inode *inode, struct file *file) -- 2.35.1

2 1

[PATCH v9 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 14 Jul '22

14 Jul '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v8: - Remove unnecessary comments and blank line. - Use devm_add_action_or_reset to ease error handling when dev probe. - Remove .remove callback in plaform driver. Changes since v7: - Optimize the code logic, such as the return value and position of a function. - Use devm_pm_runtime_enable() instead of pm_runtime_enable(). - Use pm_runtime_resume_and_get() instead of pm_runtime_get_sync(). Changes since v6: - The Makefile and Kconfig updates are moved into the driver patches so that can keep independence of each patch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 2 +- sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 255 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3000 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 57 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 16 files changed, 5942 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.18.0

2 11

[PATCH 1/2] ASoC: codecs: add support for ES8326
by Zhu Ning 14 Jul '22

14 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 761 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 ++++++++++ 4 files changed, 955 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..728623128da1 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,761 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DECTECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + snd_soc_component_write(comp, ES8326_ANA_MICBIAS_1B, 0x70); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + snd_soc_component_write(comp, ES8326_ANA_MICBIAS_1B, 0x7c); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + + return IRQ_HANDLED; +} + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PAGGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRVIER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRVIER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRVIER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRVIER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DECTECT_FB, ES8326_HP_DECTECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + break; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + if (snd_soc_component_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(es8326->mclk); + } else { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_RATES SNDRV_PCM_RATE_8000_96000 + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PAGGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + u8 reg; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) + snd_soc_jack_report(es8326->jack, 0, SND_JACK_BTN_0); + + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..23a099add4d8 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.c -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PAGGAIN_23 0x23 +#define ES8326_HP_DRVIER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DECTECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PAGGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DECTECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif \ No newline at end of file -- 2.36.1

3 4

amidi: restore space between bytes
by GitHub pull_request - opened 14 Jul '22

14 Jul '22

alsa-project/alsa-utils pull request #158 was opened from johnkeeping: Commit 9a8fcec ("amidi: add timestamp option for dump") removed the space between bytes when dumping MIDI, changing the output from: 90 45 40 80 45 00 to: 904540 804500 It seems that this was an unintentional side effect of refactoring the code to add the new timestamp output but the result is less readable than it was before. Restore the space between bytes in the same message. Fixes: 9a8fcec ("amidi: add timestamp option for dump") Signed-off-by: John Keeping <john(a)metanate.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/158 Patch URL : https://github.com/alsa-project/alsa-utils/pull/158.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: usb-audio: Use atomic_try_cmpxchg in ep_state_update
by Uros Bizjak 14 Jul '22

14 Jul '22

Use atomic_try_cmpxchg instead of atomic_cmpxchg (*ptr, old, new) == old in ep_state_update. x86 CMPXCHG instruction returns success in ZF flag, so this change saves a compare after cmpxchg (and related move instruction in front of cmpxchg). No functional change intended. Signed-off-by: Uros Bizjak <ubizjak(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index f9c921683948..0d7b73bf7945 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -133,7 +133,7 @@ static inline bool ep_state_running(struct snd_usb_endpoint *ep) static inline bool ep_state_update(struct snd_usb_endpoint *ep, int old, int new) { - return atomic_cmpxchg(&ep->state, old, new) == old; + return atomic_try_cmpxchg(&ep->state, &old, new); } /** -- 2.35.3

2 1

[PATCH] ASoC: dt-bindings: atmel-i2s: Convert to json-schema
by Ryan.Wanner＠microchip.com 14 Jul '22

14 Jul '22

From: Ryan Wanner <Ryan.Wanner(a)microchip.com> Convert atmel i2s devicetree binding to json-schema. Change file name to match json-schema naming. Signed-off-by: Ryan Wanner <Ryan.Wanner(a)microchip.com> --- .../bindings/sound/atmel,sama5d2-i2s.yaml | 83 +++++++++++++++++++ .../devicetree/bindings/sound/atmel-i2s.txt | 46 ---------- 2 files changed, 83 insertions(+), 46 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml new file mode 100644 index 000000000000..1cadc476565c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +# Copyright (C) 2022 Microchip Technology, Inc. and its subsidiaries +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sama5d2-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel I2S controller + +maintainers: + - Nicolas Ferre <nicolas.ferre(a)microchip.com> + - Alexandre Belloni <alexandre.belloni(a)bootlin.com> + - Claudiu Beznea <claudiu.beznea(a)microchip.com> + +description: + Atmel I2S (Inter-IC Sound Controller) bus is the standard + interface for connecting audio devices, such as audio codecs. + +properties: + compatible: + const: atmel,sama5d2-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + description: + Only the peripheral clock (pclk) is required. The generated clock (gclk) + and the I2S mux clock (muxclk) are optional and should only be set together, + when Master Mode is required. + + clock-names: + items: + - const: pclk + - const: gclk + - const: muxclk + minItems: 1 + + dmas: + description: + Should be one per channel name listed in the dma-names property. + maxItems: 2 + + dma-names: + items: + - const: tx + - const: rx + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + i2s@f8050000 { + compatible = "atmel,sama5d2-i2s"; + reg = <0xf8050000 0x300>; + interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(31))>, + <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(32))>; + dma-names = "tx", "rx"; + clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; + clock-names = "pclk", "gclk", "muxclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s0_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-i2s.txt b/Documentation/devicetree/bindings/sound/atmel-i2s.txt deleted file mode 100644 index 40549f496a81..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-i2s.txt +++ /dev/null @@ -1,46 +0,0 @@ -* Atmel I2S controller - -Required properties: -- compatible: Should be "atmel,sama5d2-i2s". -- reg: Should be the physical base address of the controller and the - length of memory mapped region. -- interrupts: Should contain the interrupt for the controller. -- dmas: Should be one per channel name listed in the dma-names property, - as described in atmel-dma.txt and dma.txt files. -- dma-names: Two dmas have to be defined, "tx" and "rx". - This IP also supports one shared channel for both rx and tx; - if this mode is used, one "rx-tx" name must be used. -- clocks: Must contain an entry for each entry in clock-names. - Please refer to clock-bindings.txt. -- clock-names: Should be one of each entry matching the clocks phandles list: - - "pclk" (peripheral clock) Required. - - "gclk" (generated clock) Optional (1). - - "muxclk" (I2S mux clock) Optional (1). - -Optional properties: -- pinctrl-0: Should specify pin control groups used for this controller. -- princtrl-names: Should contain only one value - "default". - - -(1) : Only the peripheral clock is required. The generated clock and the I2S - mux clock are optional and should only be set together, when Master Mode - is required. - -Example: - - i2s@f8050000 { - compatible = "atmel,sama5d2-i2s"; - reg = <0xf8050000 0x300>; - interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; - dmas = <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(31))>, - <&dma0 - (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | - AT91_XDMAC_DT_PERID(32))>; - dma-names = "tx", "rx"; - clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; - clock-names = "pclk", "gclk", "muxclk"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s0_default>; - }; -- 2.34.1

3 3

[PATCH AUTOSEL 4.9 1/4] ASoC: ops: Fix off by one in range control validation
by Sasha Levin 14 Jul '22

14 Jul '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 5871321fb4558c55bf9567052b618ff0be6b975e ] We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 90ba5521c189..4fda8c24be29 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -535,7 +535,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -556,7 +556,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.35.1

1 1

[PATCH AUTOSEL 4.14 1/5] ASoC: ops: Fix off by one in range control validation
by Sasha Levin 14 Jul '22

14 Jul '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 5871321fb4558c55bf9567052b618ff0be6b975e ] We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 0848aec1bd24..81c9ecfa7c7f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -535,7 +535,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -556,7 +556,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.35.1

1 1

[PATCH AUTOSEL 4.19 1/6] ASoC: ops: Fix off by one in range control validation
by Sasha Levin 14 Jul '22

14 Jul '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 5871321fb4558c55bf9567052b618ff0be6b975e ] We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 7a37312c8e0c..453b61b42dd9 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -530,7 +530,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -551,7 +551,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.35.1

1 1

[PATCH AUTOSEL 5.4 01/10] ASoC: ops: Fix off by one in range control validation
by Sasha Levin 14 Jul '22

14 Jul '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 5871321fb4558c55bf9567052b618ff0be6b975e ] We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 7a37312c8e0c..453b61b42dd9 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -530,7 +530,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -551,7 +551,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.35.1

1 4

[PATCH AUTOSEL 5.10 01/15] ASoC: ops: Fix off by one in range control validation
by Sasha Levin 14 Jul '22

14 Jul '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 5871321fb4558c55bf9567052b618ff0be6b975e ] We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 15bfcdbdfaa4..0f26d6c31ce5 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -517,7 +517,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -538,7 +538,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.35.1

1 6

[PATCH AUTOSEL 5.18 01/41] ASoC: ops: Fix off by one in range control validation
by Sasha Levin 14 Jul '22

14 Jul '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 5871321fb4558c55bf9567052b618ff0be6b975e ] We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e693070f51fe..d867f449d82d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -526,7 +526,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -547,7 +547,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.35.1

1 26

snd_hda_intel - HDMI missing channels / wrong mapped / no passthru
by GitHub issues - opened 14 Jul '22

14 Jul '22

alsa-project/alsa-lib issue #252 was opened from Nightwing0815: Hey Guys, I have since a few days trouble to get sound configured...more problems than on windows... I ran the alsa-info.sh script, here's the output: [https://alsa-project.org/db/?f=a95b27a60a9e38120a1d3b121ff4a93d89b1ac09](ht… I only want to configure the usuals: - card1 is microphone / headset / stereo on laptop @ full duplex, - card 0 should be HDMI stereo, HDMI digital passthru AC3 / DTS bitstreaming. I hope, you can help me with my configuration, I'm confused. Thanks in advance, ToM Issue URL : https://github.com/alsa-project/alsa-lib/issues/252 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ASoC: codecs: add support for ES8326
by Zhu Ning 14 Jul '22

14 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 759 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 ++++++++++ 4 files changed, 953 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..9b1993a5bdf3 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,759 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DETECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + snd_soc_component_write(comp, ES8326_ANA_MICBIAS_1B, 0x70); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + snd_soc_component_write(comp, ES8326_ANA_MICBIAS_1B, 0x7c); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + + return IRQ_HANDLED; +} + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_RATES SNDRV_PCM_RATE_8000_96000 + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + u8 reg; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) + snd_soc_jack_report(es8326->jack, 0, SND_JACK_BTN_0); + + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..9affd5c14407 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.c -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

1 1

[Enhancement] axfer: survive system suspend/resume while streaming
by GitHub issues - opened 14 Jul '22

14 Jul '22

alsa-project/alsa-utils issue #157 was opened from aiChaoSONG: If we suspend the system while axfer is doing playback/capture, after system resumed, axfer stopped playbak/capture with: ``` Fail to process frames: Streams pipe error PLAYBACK: Expected 4611686018427387903frames, Actual 143392frames snd_pcm_drain(): File descriptor in bad state snd_pcm_hw_free(): File descriptor in bad state ``` You can get the failure by playing in a terminal with `axfer transfer playback -Dhw:0,0 -f dat -i -vvv -c 2 /dev/zero`, and run `sudo rtcwake -m mem -s 5` in another terminal. axfer should survive system suspend/resume, just like aplay. @takaswie FYI, thank you for creating this tool. here is how aplay handle this: https://github.com/alsa-project/alsa-utils/blob/a566f8a0ed6d7ca5fd6ae2d224f… Issue URL : https://github.com/alsa-project/alsa-utils/issues/157 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: rockchip-i2s: Undo BCLK pinctrl changes
by Mark Brown 13 Jul '22

13 Jul '22

The version of the BCLK pinctrl management changes that made it into v5.19 has caused problems on some systems due to overly strict DT requirements but attempts to fix it have caused further breakage on other platforms. Just drop the changes for this release, we already have a better version queued for -next. Fixes: 26b9f2fa7b1c ("ASoC: rockchip: i2s: Fix NULL pointer dereference when pinctrl is not found") Fixes: a5450aba737d ("ASoC: rockchip: i2s: switch BCLK to GPIO") Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 163 ++++++------------------------ 1 file changed, 31 insertions(+), 132 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index c9fedf6eb2e6..4ce5d2579387 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -13,7 +13,6 @@ #include <linux/of_gpio.h> #include <linux/of_device.h> #include <linux/clk.h> -#include <linux/pinctrl/consumer.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/spinlock.h> @@ -55,40 +54,8 @@ struct rk_i2s_dev { const struct rk_i2s_pins *pins; unsigned int bclk_ratio; spinlock_t lock; /* tx/rx lock */ - struct pinctrl *pinctrl; - struct pinctrl_state *bclk_on; - struct pinctrl_state *bclk_off; }; -static int i2s_pinctrl_select_bclk_on(struct rk_i2s_dev *i2s) -{ - int ret = 0; - - if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_on)) - ret = pinctrl_select_state(i2s->pinctrl, - i2s->bclk_on); - - if (ret) - dev_err(i2s->dev, "bclk enable failed %d\n", ret); - - return ret; -} - -static int i2s_pinctrl_select_bclk_off(struct rk_i2s_dev *i2s) -{ - - int ret = 0; - - if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_off)) - ret = pinctrl_select_state(i2s->pinctrl, - i2s->bclk_off); - - if (ret) - dev_err(i2s->dev, "bclk disable failed %d\n", ret); - - return ret; -} - static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); @@ -125,49 +92,38 @@ static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) return snd_soc_dai_get_drvdata(dai); } -static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; - int ret = 0; spin_lock(&i2s->lock); if (on) { - ret = regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); - if (ret < 0) - goto end; + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); - ret = regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); - if (ret < 0) - goto end; + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); i2s->tx_start = true; } else { i2s->tx_start = false; - ret = regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - if (ret < 0) - goto end; + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); if (!i2s->rx_start) { - ret = regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); - if (ret < 0) - goto end; + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); udelay(150); - ret = regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); - if (ret < 0) - goto end; + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); regmap_read(i2s->regmap, I2S_CLR, &val); @@ -182,57 +138,44 @@ static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } -end: spin_unlock(&i2s->lock); - if (ret < 0) - dev_err(i2s->dev, "lrclk update failed\n"); - - return ret; } -static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; - int ret = 0; spin_lock(&i2s->lock); if (on) { - ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); - if (ret < 0) - goto end; - ret = regmap_update_bits(i2s->regmap, I2S_XFER, + regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_START | I2S_XFER_RXS_START); - if (ret < 0) - goto end; i2s->rx_start = true; } else { i2s->rx_start = false; - ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - if (ret < 0) - goto end; if (!i2s->tx_start) { - ret = regmap_update_bits(i2s->regmap, I2S_XFER, + regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); - if (ret < 0) - goto end; + udelay(150); - ret = regmap_update_bits(i2s->regmap, I2S_CLR, + regmap_update_bits(i2s->regmap, I2S_CLR, I2S_CLR_TXC | I2S_CLR_RXC, I2S_CLR_TXC | I2S_CLR_RXC); - if (ret < 0) - goto end; + regmap_read(i2s->regmap, I2S_CLR, &val); + /* Should wait for clear operation to finish */ while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); @@ -244,12 +187,7 @@ static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } -end: spin_unlock(&i2s->lock); - if (ret < 0) - dev_err(i2s->dev, "lrclk update failed\n"); - - return ret; } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -487,26 +425,17 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ret = rockchip_snd_rxctrl(i2s, 1); + rockchip_snd_rxctrl(i2s, 1); else - ret = rockchip_snd_txctrl(i2s, 1); - /* Do not turn on bclk if lrclk open fails. */ - if (ret < 0) - return ret; - i2s_pinctrl_select_bclk_on(i2s); + rockchip_snd_txctrl(i2s, 1); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (!i2s->tx_start) - i2s_pinctrl_select_bclk_off(i2s); - ret = rockchip_snd_rxctrl(i2s, 0); - } else { - if (!i2s->rx_start) - i2s_pinctrl_select_bclk_off(i2s); - ret = rockchip_snd_txctrl(i2s, 0); - } + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_snd_rxctrl(i2s, 0); + else + rockchip_snd_txctrl(i2s, 0); break; default: ret = -EINVAL; @@ -807,36 +736,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } i2s->bclk_ratio = 64; - i2s->pinctrl = devm_pinctrl_get(&pdev->dev); - if (IS_ERR(i2s->pinctrl)) { - dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); - ret = PTR_ERR(i2s->pinctrl); - goto err_clk; - } - - i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, - "bclk_on"); - if (IS_ERR_OR_NULL(i2s->bclk_on)) - dev_err(&pdev->dev, "failed to find i2s default state\n"); - else - dev_dbg(&pdev->dev, "find i2s bclk state\n"); - - i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, - "bclk_off"); - if (IS_ERR_OR_NULL(i2s->bclk_off)) - dev_err(&pdev->dev, "failed to find i2s gpio state\n"); - else - dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); - - i2s_pinctrl_select_bclk_off(i2s); - - i2s->playback_dma_data.addr = res->start + I2S_TXDR; - i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->playback_dma_data.maxburst = 4; - - i2s->capture_dma_data.addr = res->start + I2S_RXDR; - i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->capture_dma_data.maxburst = 4; dev_set_drvdata(&pdev->dev, i2s); -- 2.30.2

2 4

[RESEND PATCH v3 0/2] ASoC/SoundWire: Intel: add sdw BE dai trigger
by Bard Liao 13 Jul '22

13 Jul '22

For SOF IPC4, we need to set pipeline state in BE DAI trigger. @Mark, resending this series in case it is lost in your mailbox. v2: - Change "#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)" to "if (IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE))" v3: - Change the multi-line comment style in "soundwire: Intel: add trigger callback" patch and take Vinod's Ack. Bard Liao (2): soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback drivers/soundwire/intel.c | 9 +++++++++ include/linux/soundwire/sdw_intel.h | 1 + sound/soc/sof/intel/hda-dai.c | 14 +++++++++++--- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 1 + 5 files changed, 23 insertions(+), 4 deletions(-) -- 2.17.1

2 3

[PATCH] soundwire: qcom: Update error prints to debug prints
by Srinivasa Rao Mandadapu 13 Jul '22

13 Jul '22

Upadte error prints to debug prints to avoid redundant logging in kernel boot time, as these prints are informative prints in irq handler. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- drivers/soundwire/qcom.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970e..a4293d0 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -573,11 +573,10 @@ static irqreturn_t qcom_swrm_irq_handler(int irq, void *dev_id) break; case SWRM_INTERRUPT_STATUS_NEW_SLAVE_ATTACHED: case SWRM_INTERRUPT_STATUS_CHANGE_ENUM_SLAVE_STATUS: - dev_err_ratelimited(swrm->dev, "%s: SWR new slave attached\n", - __func__); + dev_dbg(swrm->dev, "%s: SWR new slave attached\n", __func__); swrm->reg_read(swrm, SWRM_MCP_SLV_STATUS, &slave_status); if (swrm->slave_status == slave_status) { - dev_err(swrm->dev, "Slave status not changed %x\n", + dev_dbg(swrm->dev, "Slave status not changed %x\n", slave_status); } else { qcom_swrm_get_device_status(swrm); -- 2.7.4

4 6

[PATCH -next] ASoC: amd: fix Jadeite kconfig warning and build errors
by Randy Dunlap 13 Jul '22

13 Jul '22

Since SND_SOC_ES8316 has a hard dependency on I2C and since 'select' does not follow any dependency chains, SND_SOC_AMD_ST_ES8336_MACH also needs to have a hard dependency on I2C. Fixes a kconfig warning and subsequent build errors: WARNING: unmet direct dependencies detected for SND_SOC_ES8316 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=n] Selected by [y]: - SND_SOC_AMD_ST_ES8336_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_ACP [=y] && ACPI [=y] && (I2C [=n] || COMPILE_TEST [=y]) sound/soc/codecs/es8316.c:866:1: warning: data definition has no type or storage class 866 | module_i2c_driver(es8316_i2c_driver); sound/soc/codecs/es8316.c:866:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/es8316.c:866:1: warning: parameter names (without types) in function declaration sound/soc/codecs/es8316.c:857:26: warning: ‘es8316_i2c_driver’ defined but not used [-Wunused-variable] 857 | static struct i2c_driver es8316_i2c_driver = { Fixes: f94fa8405801 ("ASoC: amd: enable machine driver build for Jadeite platform") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -28,7 +28,7 @@ config SND_SOC_AMD_ST_ES8336_MACH select SND_SOC_ACPI if ACPI select SND_SOC_ES8316 depends on SND_SOC_AMD_ACP && ACPI - depends on I2C || COMPILE_TEST + depends on I2C help This option enables machine driver for Jadeite platform using es8336 codec.

2 1

[PATCH] ASoC: amd: Fix error pointer dereference
by Dan Carpenter 13 Jul '22

13 Jul '22

The "gpio_pa" pointer is an error pointer, there is no need to try put it. Calling gpiod_put() on it will lead to an error pointer dereference. Fixes: 02527c3f2300 ("ASoC: amd: add Machine driver for Jadeite platform") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/acp-es8336.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index d501618b78f6..2fe8df86053a 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -212,7 +212,6 @@ static int st_es8336_late_probe(struct snd_soc_card *card) if (IS_ERR(gpio_pa)) { ret = dev_err_probe(card->dev, PTR_ERR(gpio_pa), "could not get pa-enable GPIO\n"); - gpiod_put(gpio_pa); put_device(codec_dev); return ret; } -- 2.35.1

2 1

[PATCH] ASoc: audio-graph-card2: Fix refcount leak bug in __graph_get_type()
by Liang He 13 Jul '22

13 Jul '22

We should call of_node_put() for the reference before its replacement as it returned by of_get_parent() which has increased the refcount. Besides, we should also call of_node_put() before return. Fixes: c8c74939f791 ("ASoC: audio-graph-card2: add Multi CPU/Codec support") Signed-off-by: Liang He <windhl(a)126.com> --- sound/soc/generic/audio-graph-card2.c | 35 +++++++++++++++++++-------- 1 file changed, 25 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 77ac4051b827..4f6bae73bd5e 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -229,7 +229,8 @@ enum graph_type { static enum graph_type __graph_get_type(struct device_node *lnk) { - struct device_node *np; + struct device_node *np, *parent_np; + enum graph_type ret; /* * target { @@ -240,19 +241,33 @@ static enum graph_type __graph_get_type(struct device_node *lnk) * }; */ np = of_get_parent(lnk); - if (of_node_name_eq(np, "ports")) - np = of_get_parent(np); + if (of_node_name_eq(np, "ports")) { + parent_np = of_get_parent(np); + of_node_put(np); + np = parent_np; + } - if (of_node_name_eq(np, GRAPH_NODENAME_MULTI)) - return GRAPH_MULTI; + if (of_node_name_eq(np, GRAPH_NODENAME_MULTI)) { + ret = GRAPH_MULTI; + goto out_put; + } - if (of_node_name_eq(np, GRAPH_NODENAME_DPCM)) - return GRAPH_DPCM; + if (of_node_name_eq(np, GRAPH_NODENAME_DPCM)) { + ret = GRAPH_DPCM; + goto out_put; + } - if (of_node_name_eq(np, GRAPH_NODENAME_C2C)) - return GRAPH_C2C; + if (of_node_name_eq(np, GRAPH_NODENAME_C2C)) { + ret = GRAPH_C2C; + goto out_put; + } + + ret = GRAPH_NORMAL; + +out_put: + of_node_put(np); + return ret; - return GRAPH_NORMAL; } static enum graph_type graph_get_type(struct asoc_simple_priv *priv, -- 2.25.1

2 1

[PATCH] ASoC: mt6359: Fix refcount leak bug
by Liang He 13 Jul '22

13 Jul '22

In mt6359_parse_dt() and mt6359_accdet_parse_dt(), we should call of_node_put() for the reference returned by of_get_child_by_name() which has increased the refcount. Fixes: 683530285316 ("ASoC: mt6359: fix failed to parse DT properties") Fixes: eef07b9e0925 ("ASoC: mediatek: mt6359: add MT6359 accdet jack driver") Signed-off-by: Liang He <windhl(a)126.com> --- sound/soc/codecs/mt6359-accdet.c | 1 + sound/soc/codecs/mt6359.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index 6d3d170144a0..c190628e2905 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -675,6 +675,7 @@ static int mt6359_accdet_parse_dt(struct mt6359_accdet *priv) sizeof(struct three_key_threshold)); } + of_node_put(node); dev_warn(priv->dev, "accdet caps=%x\n", priv->caps); return 0; diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 23709b180409..c9a453ce8a2a 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2778,6 +2778,7 @@ static int mt6359_parse_dt(struct mt6359_priv *priv) ret = of_property_read_u32(np, "mediatek,mic-type-2", &priv->mux_select[MUX_MIC_TYPE_2]); + of_node_put(np); if (ret) { dev_info(priv->dev, "%s() failed to read mic-type-2, use default (%d)\n", -- 2.25.1

2 1

[PATCH 1/2] soundwire: bus: remove use of __func__ in dev_dbg
by Bard Liao 13 Jul '22

13 Jul '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH <gregkh(a)linuxfoundation.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 8970f8560766..2772973eebb1 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -827,13 +827,13 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, mutex_lock(&bus->bus_lock); dev_vdbg(bus->dev, - "%s: changing status slave %d status %d new status %d\n", - __func__, slave->dev_num, slave->status, status); + "changing status slave %d status %d new status %d\n", + slave->dev_num, slave->status, status); if (status == SDW_SLAVE_UNATTACHED) { dev_dbg(&slave->dev, - "%s: initializing enumeration and init completion for Slave %d\n", - __func__, slave->dev_num); + "initializing enumeration and init completion for Slave %d\n", + slave->dev_num); init_completion(&slave->enumeration_complete); init_completion(&slave->initialization_complete); @@ -841,8 +841,8 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, } else if ((status == SDW_SLAVE_ATTACHED) && (slave->status == SDW_SLAVE_UNATTACHED)) { dev_dbg(&slave->dev, - "%s: signaling enumeration completion for Slave %d\n", - __func__, slave->dev_num); + "signaling enumeration completion for Slave %d\n", + slave->dev_num); complete(&slave->enumeration_complete); } @@ -1849,8 +1849,8 @@ int sdw_handle_slave_status(struct sdw_bus *bus, "Update Slave status failed:%d\n", ret); if (attached_initializing) { dev_dbg(&slave->dev, - "%s: signaling initialization completion for Slave %d\n", - __func__, slave->dev_num); + "signaling initialization completion for Slave %d\n", + slave->dev_num); complete(&slave->initialization_complete); -- 2.25.1

1 1

[PATCH 0/3] soundwire: allocate device_number with IDA
by Bard Liao 13 Jul '22

13 Jul '22

The device_number is currently allocated in the scope of each bus and does not need to be unique at the system level. This leads e.g. on Dell devices with three or four Device1 on different bus segments. To make the device_number unique at the system level, and unified with the HDaudio/iDISP SDI values, this PRs allocates the dev_number with an IDA restricted between 4 and 11 (inclusive). Pierre-Louis Bossart (3): soundwire: bus: rename sdw_ida as sdw_bus_ida soundwire: bus: allow device number to be unique at system level soundwire: intel: set dev_num_ida_min drivers/soundwire/bus.c | 29 ++++++++++++++++++++--------- drivers/soundwire/intel.c | 4 ++++ include/linux/soundwire/sdw.h | 4 ++++ 3 files changed, 28 insertions(+), 9 deletions(-) -- 2.25.1

1 3

[bug report] ASoC: amd: acp: ACP code generic to support newer platforms
by Dan Carpenter 13 Jul '22

13 Jul '22

Hello V sujith kumar Reddy, This is a semi-automatic email about new static checker warnings. The patch b24484c18b10: "ASoC: amd: acp: ACP code generic to support newer platforms" from Jul 7, 2022, leads to the following Smatch complaint: sound/soc/amd/acp/acp-platform.c:99 i2s_irq_handler() warn: variable dereferenced before check 'adata' (see line 94) sound/soc/amd/acp/acp-platform.c 93 struct acp_dev_data *adata = data; 94 struct acp_resource *rsrc = adata->rsrc; ^^^^^^^^^^^ The patch adds a new dereference 95 struct acp_stream *stream; 96 u16 i2s_flag = 0; 97 u32 val, val1, i; 98 99 if (!adata) ^^^^^^ The old code assumed that "adata" could be NULL but hopefully that check can be removed? 100 return IRQ_NONE; 101 regards, dan carpenter

1 0

snd-aloop working no more on 1.2.7.1
by GitHub issues - edited 13 Jul '22

13 Jul '22

alsa-project/alsa-lib issue #251 was edited from tramchamploo: ``` pcm.er4s { # Direct hardware access type hw # Loopback card name # # Has to match "id" in the options of the snd-aloop module card "Loopback" # Loopback device ID device 0 # Number of audio channels # # Has to match the number of channels in music player app # and in the CamillaDSP input configuration channels 2 # Format of audio stream # # Has to match the format defined in the # of the CamillaDSP input configuration format "FLOAT_LE" # Sampling-rate of audio stream # # Has to match the sampling-rate defined in the # CamillaDSP configuration rate 48000 } ctl.er4s { type hw card "Loopback" } ``` This is the .asoundrc I'm using, it worked before this update. Now aplay -L doesn't show this device. Using mpd to play to the Loopback gets `snd_pcm_hw_params() failed: Invalid argument`. Here's my mpd config: ``` audio_output { type "alsa" name "Etymotic ER4S" device "er4s" format "48000:f:2" mixer_type "none" replay_gain_handler "none" auto_resample "no" auto_format "no" } resampler { plugin "soxr" quality "custom" precision "28" phase_response "50" passband_end "91" stopband_begin "100" attenu ``` Btw, alsa should really have a detailed specification on asoundrc file. The value of the format option I specify is from a dsp project, otherwise I have no idea what to write here. And according to [https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gaa14b7f2… it's in a totally different format. Issue URL : https://github.com/alsa-project/alsa-lib/issues/251 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-aloop working no more on 1.2.7.1
by GitHub issues - edited 13 Jul '22

13 Jul '22

alsa-project/alsa-lib issue #251 was edited from tramchamploo: ``` pcm.er4s { # Direct hardware access type hw # Loopback card name # # Has to match "id" in the options of the snd-aloop module card "Loopback" # Loopback device ID device 0 # Number of audio channels # # Has to match the number of channels in music player app # and in the CamillaDSP input configuration channels 2 # Format of audio stream # # Has to match the format defined in the # of the CamillaDSP input configuration format "FLOAT_LE" # Sampling-rate of audio stream # # Has to match the sampling-rate defined in the # CamillaDSP configuration rate 48000 } ctl.er4s { type hw card "Loopback" } ``` This is the .asoundrc I'm using, it worked before this update. Now aplay -L doesn't show this device. Using mpd to play to the Loopback gets `snd_pcm_hw_params() failed: Invalid argument`. Btw, alsa should really have a detailed specification on asoundrc file. The value of the format option I specify is from a dsp project, otherwise I have no idea what to write here. And according to [https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gaa14b7f2… it's in a totally different format. Issue URL : https://github.com/alsa-project/alsa-lib/issues/251 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-aloop working no more on 1.2.7.1
by GitHub issues - opened 13 Jul '22

13 Jul '22

alsa-project/alsa-lib issue #251 was opened from tramchamploo: `# # Place your global alsa-lib configuration here... # # pcm.er4s { # Direct hardware access type hw # Loopback card name # # Has to match "id" in the options of the snd-aloop module card "Loopback" # Loopback device ID device 0 # Number of audio channels # # Has to match the number of channels in music player app # and in the CamillaDSP input configuration channels 2 # Format of audio stream # # Has to match the format defined in the # of the CamillaDSP input configuration format "FLOAT_LE" # Sampling-rate of audio stream # # Has to match the sampling-rate defined in the # CamillaDSP configuration rate 48000 } ctl.er4s { type hw card "Loopback" } ` This is the .asoundrc I'm using, it worked before this update. Now aplay -L doesn't show this device. Using mpd to play to the Loopback gets `snd_pcm_hw_params() failed: Invalid argument`. Btw, alsa should really have a detailed specification on asoundrc file. The value of the format option I specify is from a dsp project, otherwise I have no idea what to write here. And according to [https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gaa14b7f2… it's in a totally different format. Issue URL : https://github.com/alsa-project/alsa-lib/issues/251 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/7] ALSA: kernel doc fixes
by Takashi Iwai 13 Jul '22

13 Jul '22

Hi, this is a patch series to address minor warnings at processing kernel docs. Takashi === Takashi Iwai (7): ALSA: compress: Enable kernel doc markers for some functions ALSA: pcm: Fix missing return value comments for kernel docs ALSA: dmaengine: Fix missing return value comments for kernel docs ALSA: compress: Fix kernel doc warnings ALSA: control: Fix missing return value comments for kernel docs ALSA: memalloc: Fix missing return value comments for kernel docs ALSA: core: Fix missing return value comments for kernel docs include/sound/dmaengine_pcm.h | 2 ++ include/sound/pcm.h | 50 +++++++++++++++++++++++++-- include/uapi/sound/compress_offload.h | 2 +- include/uapi/sound/compress_params.h | 6 ++-- sound/core/compress_offload.c | 9 +++-- sound/core/control.c | 16 +++++++-- sound/core/device.c | 2 ++ sound/core/info.c | 2 ++ sound/core/init.c | 10 ++++++ sound/core/isadma.c | 3 +- sound/core/memalloc.c | 10 +++++- sound/core/pcm.c | 4 +++ sound/core/pcm_dmaengine.c | 30 ++++++++++------ sound/core/pcm_memory.c | 4 +++ sound/core/pcm_native.c | 6 ++++ sound/core/vmaster.c | 3 +- 16 files changed, 136 insertions(+), 23 deletions(-) -- 2.35.3

1 7

[PATCH] ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop
by Meng Tang 13 Jul '22

13 Jul '22

The headset on this machine is not defined, after applying the quirk ALC256_FIXUP_ASUS_HEADSET_MIC, the headset-mic works well Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 91cd58849ee8..383a814b8539 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9369,6 +9369,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc221
by Meng Tang 13 Jul '22

13 Jul '22

On a HP 288 Pro G2 MT (X9W02AV), the front mic could not be detected. In order to get it working, the pin configuration needs to be set correctly, and the ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE fixup needs to be applied. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34139c26795f..91cd58849ee8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6901,6 +6901,7 @@ enum { ALC298_FIXUP_LENOVO_SPK_VOLUME, ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, ALC269_FIXUP_ATIV_BOOK_8, + ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE, ALC221_FIXUP_HP_MIC_NO_PRESENCE, ALC256_FIXUP_ASUS_HEADSET_MODE, ALC256_FIXUP_ASUS_MIC, @@ -7837,6 +7838,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_NO_SHUTUP }, + [ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01813030 }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC221_FIXUP_HP_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9012,6 +9023,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2b5e, "HP 288 Pro G2 MT", ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP machines
by Jeremy Szu 13 Jul '22

13 Jul '22

* The HP ProBook 440/450 G9 and EliteBook 640/650 G9 have multiple motherboard design and they are using different subsystem ID of audio codec. Add the same quirk for other MBs. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 007dd8b5e1f2..483c5f49af5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9096,6 +9096,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.25.1

2 1

[PATCH v2 1/4] ALSA: hda: cs35l41: Don't dereference fwnode handle
by Andy Shevchenko 13 Jul '22

13 Jul '22

Use acpi_fwnode_handle() instead of dereferencing an fwnode handle directly, which is a better coding practice. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- v2: new change sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 49b25432a9f5..1a1afa0725e0 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -347,7 +347,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i /* To use the same release code for all laptop variants we can't use devm_ version of * gpiod_get here, as CLSA010* don't have a fully functional bios with an _DSD node */ - cs35l41->reset_gpio = fwnode_gpiod_get_index(&adev->fwnode, "reset", cs35l41->index, + cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(adev), "reset", cs35l41->index, GPIOD_OUT_LOW, "cs35l41-reset"); property = "cirrus,speaker-position"; -- 2.35.1

2 4

[PATCH 0/4] Extend ipc stream parameters sent to DSP
by Daniel Baluta 12 Jul '22

12 Jul '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We need a way to send extra parameters to DSP firmware. In order to do this, we introduce ext_data array at the end of ipc_stream_params. With this new addition we can send compress parameters. This requires SOF ABI bump. Daniel Baluta (4): ASoC: SOF: compress: Dynamically allocate pcm params struct ASoC: SOF: Copy compress parameters into extended data ASoC: SOF: compress: Prevent current kernel running with older FW uapi: sof: abi: Bump SOF ABI for ext_data_length include/sound/sof/stream.h | 6 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/compress.c | 73 ++++++++++++++++++++++++------------ 3 files changed, 54 insertions(+), 27 deletions(-) -- 2.27.0

2 5

[PATCH 0/2] ASoC: SOF: Intel: hda-dai: No need to decouple host/link DMA twice
by Peter Ujfalusi 12 Jul '22

12 Jul '22

Hi, There is no need to decouple a decoupled stream twice. Keep the decoupling in hda_link_stream_assign() only as it is going to be executed in all cases. Drop the outdated comment from hda_link_dma_hw_params() as well since the code has changed around it. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: Intel: hda-dai: Drop misleading comment regarding dma_data ASoC: SOF: Intel: hda-dai: Do snd_hdac_ext_stream_decouple() only once sound/soc/sof/intel/hda-dai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) -- 2.37.0

2 3

[PATCH] ASoC: SOF: sof-client-probes: Only load the driver if IPC3 is used
by Peter Ujfalusi 12 Jul '22

12 Jul '22

The current implementation of probes only supports IPC3 and should not be loaded for other IPC implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/sof-client-probes.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 1f1ea93a7fbf..e7e78d1a8ec3 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -694,6 +694,10 @@ static int sof_probes_client_probe(struct auxiliary_device *auxdev, if (!sof_probes_enabled) return -ENXIO; + /* only ipc3 is supported */ + if (sof_client_get_ipc_type(cdev) != SOF_IPC) + return -ENXIO; + if (!dev->platform_data) { dev_err(dev, "missing platform data\n"); return -ENODEV; -- 2.37.0

2 1

[PATCH] ASoC: SOF: ipc3-topology: Prevent double freeing of ipc_control_data via load_bytes
by Peter Ujfalusi 12 Jul '22

12 Jul '22

We have sanity checks for byte controls and if any of the fail the locally allocated scontrol->ipc_control_data is freed up, but not set to NULL. On a rollback path of the error the higher level code will also try to free the scontrol->ipc_control_data which will eventually going to lead to memory corruption as double freeing memory is not a good thing. Fixes: b5cee8feb1d4 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 9448d5338423..b2cc046b9f60 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1644,6 +1644,7 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ return 0; err: kfree(scontrol->ipc_control_data); + scontrol->ipc_control_data = NULL; return ret; } -- 2.37.0

2 1

[PATCH 0/3] ASoC: SOF: Intel: Do not process IPC reply before firmware boot
by Peter Ujfalusi 12 Jul '22

12 Jul '22

Hi, By mistake a developer managed to create a 'corrupted' IPC4 firmware image which loaded fine to the DSP and after boot it sent an IPC reply before we would have received the FW_READY message. It turned out that the image was an IPC3 firmware and the IPC reply was the IPC3 FW_READY notification message which got understood as an IPC4 reply message due to the difference between the two IPC mechanism. This caused a NULL pointer dereference since the reply memory will be allocated after the FW_READY message. To make sure this will not bite again, skip any spurious reply messages before the FW_READY. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: cnl: Do not process IPC reply before firmware boot ASoC: SOF: Intel: hda-ipc: Do not process IPC reply before firmware boot ASoC: SOF: Intel: mtl: Do not process IPC reply before firmware boot sound/soc/sof/intel/cnl.c | 37 +++++++++++++++++++++------------ sound/soc/sof/intel/hda-ipc.c | 39 ++++++++++++++++++++++------------- sound/soc/sof/intel/mtl.c | 20 +++++++++++------- 3 files changed, 62 insertions(+), 34 deletions(-) -- 2.37.0

2 4

alsalib assertion using plug+dmix
by GitHub issues - opened 12 Jul '22

12 Jul '22

alsa-project/alsa-plugins issue #46 was opened from npitre: The espeak program is crashing randomly here. I've been able to isolate the issue in a small test program demonstrating the crash 100% of the time, ending in: ``` alsa_quickcrash: pcm.c:3236: snd_pcm_area_copy: Assertion `src < dst || src >= dst + bytes' failed. Aborted (core dumped) ``` Here's the test code: ``` /* to compile: gcc -O2 -Wall -o alsa_quickcrash alsa_quickcrash.c -lasound */ #include <alsa/asoundlib.h> #include <stdio.h> #include <string.h> #define DEVICE "test_dummy" #define RATE 22050 #define CHANNELS 1 #define FORMAT SND_PCM_FORMAT_S16_LE #define BUF_MS 60 #define DO(x) if (x < 0) { fprintf(stderr, "%s failed\n", #x); exit(1); } int main() { unsigned int rate = RATE; snd_pcm_uframes_t frames = RATE * BUF_MS / 1000; snd_pcm_t *handle; snd_pcm_hw_params_t *params; DO(snd_pcm_hw_params_malloc(&params)); DO(snd_pcm_open(&handle, DEVICE, SND_PCM_STREAM_PLAYBACK, 0)); DO(snd_pcm_hw_params_any(handle, params)); DO(snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED)); DO(snd_pcm_hw_params_set_format(handle, params, FORMAT)); DO(snd_pcm_hw_params_set_rate_near(handle, params, &rate, 0)); DO(snd_pcm_hw_params_set_channels(handle, params, CHANNELS)); DO(snd_pcm_hw_params_set_buffer_size_near(handle, params, &frames)); DO(snd_pcm_hw_params(handle, params)); DO(snd_pcm_prepare(handle)); printf("PCM config dump:\n"); snd_output_t *output = NULL; DO(snd_output_stdio_attach(&output, stdout, 0)); DO(snd_pcm_dump(handle, output)); printf("\n"); printf("frames = %ld\n", (long)frames); short data[frames * CHANNELS]; memset(data, 0, sizeof(data)); printf("writing whole buffer\n"); DO(snd_pcm_writei(handle, data, frames)); printf("writing whole buffer\n"); DO(snd_pcm_writei(handle, data, frames)); printf("writing whole buffer minus 1 frame\n"); DO(snd_pcm_writei(handle, data, frames - 1)); printf("draining audio\n"); DO(snd_pcm_drain(handle)); printf("done\n"); DO(snd_pcm_close(handle)); return 0; } ``` Here's the corresponding .asoundrc content: ``` pcm.test_dummy { type plug slave.pcm "mix_dummy" } pcm.mix_dummy { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0666 slave { pcm "hw:0" period_size 1024 buffer_size 4096 format "S16_LE" rate 48000 } } ``` And here's the complete test output: ``` PCM config dump: Plug PCM: Rate conversion PCM (48000, sformat=S16_LE) Converter: linear-interpolation Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 22050 exact rate : 22050 (22050/1) msbits : 16 buffer_size : 1411 period_size : 470 period_time : 21333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 470 period_event : 0 start_threshold : 1 stop_threshold : 1411 silence_threshold: 0 silence_size : 0 boundary : 3177289537109884928 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 1024 period_time : 21333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 2 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 1024 period_time : 21333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 2 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Hardware PCM card 1 'HDA Intel PCH' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 4096 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 2734916609 frames = 1411 writing whole buffer writing whole buffer writing whole buffer minus 1 frame draining audio alsa_quickcrash: pcm.c:3236: snd_pcm_area_copy: Assertion `src < dst || src >= dst + bytes' failed. Aborted (core dumped) ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/46 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ASoC: soc-pcm: demote warnings on non-atomic BE connection
by Pierre-Louis Bossart 12 Jul '22

12 Jul '22

When an FE, typically non-atomic, is connected to an atomic BE, we force the BE as non-atomic. There's no reason to throw a warning, this is a perfectly fine configuration and a conversion that's required by-design. This removes the unconditional warnings such as [ 12.054213] iDisp1: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.074693] iDisp2: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.096612] iDisp3: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.118637] iDisp4: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.140660] dmic01: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.147521] dmic16k: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic and demotes them to dev_dbg(), as suggested in review comments. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..5b99bf2dbd08 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1209,8 +1209,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, return -EINVAL; } if (fe_substream->pcm->nonatomic && !be_substream->pcm->nonatomic) { - dev_warn(be->dev, "%s: FE is nonatomic but BE is not, forcing BE as nonatomic\n", - __func__); + dev_dbg(be->dev, "FE is nonatomic but BE is not, forcing BE as nonatomic\n"); be_substream->pcm->nonatomic = 1; } -- 2.34.1

2 1

[PATCH] ASoC: SOF: topology: remove unused variable
by Peter Ujfalusi 12 Jul '22

12 Jul '22

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> 'ret' is never used. Remove it and return 0 instead. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 7e54eb1bf77b..9273a70fec25 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1419,7 +1419,6 @@ static int sof_widget_unload(struct snd_soc_component *scomp, struct soc_bytes_ext *sbe; struct snd_sof_dai *dai; struct soc_enum *se; - int ret = 0; int i; swidget = dobj->private; @@ -1480,7 +1479,7 @@ static int sof_widget_unload(struct snd_soc_component *scomp, list_del(&swidget->list); kfree(swidget); - return ret; + return 0; } /* -- 2.37.0

2 1

[PATCH 0/2] ASoC: SOF: Intel: hda: Use cold/purge boot after firmware crash
by Peter Ujfalusi 12 Jul '22

12 Jul '22

Hi, In case of a firmware crash we force the DSP to be powered down and rebooted. To make sure that the next boot is going to be clean, force the boot process to skip the IMR booting and re-download the firmware. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: Intel: hda: Introduce skip_imr_boot flag ASoC: SOF: Intel: hda: Skip IMR boot after a firmware crash or boot failure sound/soc/sof/intel/hda-dsp.c | 12 ++++++++++++ sound/soc/sof/intel/hda-loader.c | 10 ++++++---- sound/soc/sof/intel/hda.h | 1 + 3 files changed, 19 insertions(+), 4 deletions(-) -- 2.37.0

2 3

Add profile for Behringer UMC202HD
by GitHub pull_request - opened 12 Jul '22

12 Jul '22

alsa-project/alsa-ucm-conf pull request #188 was opened from Eiseiche: Edited #128 for UMC202HD support (thx @brndd for initial work). [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9094061/… [spa-acp-tool.txt](https://github.com/alsa-project/alsa-ucm-conf/files/90940… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/188 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/188.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: Sound not working after commit bbf7d3b1c4f40eb02dd1dffb500ba00b0bff0303 on Amlogic A311D device
by Alex Natalsson 12 Jul '22

12 Jul '22

> Wow. So there's already a problem with a config and we made it worse... > > Can you try with this hack attached, just to see what causes the kernel > oops? Thanks! Hello friends) I modified your hack a little. And I don't understanding what is fe and be, but I see what problem apearing due to be_substream points to 0. The "if (!fe_substream->pcm->nonatomic && be_substream->pcm->nonatomic) {" at result gives the error... * I am sorry what delete recipients in the previous send((

4 7

NetBSD/OpenBSD/FreeBSD build support
by GitHub pull_request - opened 12 Jul '22

12 Jul '22

alsa-project/alsa-lib pull request #250 was opened from jg1uaa: to build alsa-lib on *BSD, small fix is required. (reference: FreeBSD/NetBSD ports diff) There is no problem to compile modified code on Linux(Debian-11), FreeBSD-13.1, NetBSD-current and OpenBSD-7.1. Request URL : https://github.com/alsa-project/alsa-lib/pull/250 Patch URL : https://github.com/alsa-project/alsa-lib/pull/250.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: rockchip: i2s: Fix NULL pointer dereference when pinctrl is not found
by Alexandru Elisei 12 Jul '22

12 Jul '22

Commit a5450aba737d ("ASoC: rockchip: i2s: switch BCLK to GPIO") switched BCLK to GPIO functions when probing the i2s bus interface, but missed adding a check for when devm_pinctrl_get() returns an error. This can lead to the following NULL pointer dereference on a rockpro64-v2 if there are no "pinctrl" properties in the i2s device tree node: [ 0.658381] rockchip-i2s ff880000.i2s: failed to find i2s default state [ 0.658993] rockchip-i2s ff880000.i2s: failed to find i2s gpio state [ 0.660072] rockchip-i2s ff890000.i2s: failed to find i2s default state [ 0.660670] rockchip-i2s ff890000.i2s: failed to find i2s gpio state [ 0.661716] rockchip-i2s ff8a0000.i2s: failed to find i2s pinctrl [ 0.662276] Unable to handle kernel NULL pointer dereference at virtual address 0000000000000005 [ 0.663061] Mem abort info: [ 0.663317] ESR = 0x0000000096000004 [ 0.663658] EC = 0x25: DABT (current EL), IL = 32 bits [ 0.664136] SET = 0, FnV = 0 [ 0.664171] mmc2: SDHCI controller on fe330000.mmc [fe330000.mmc] using ADMA [ 0.664409] EA = 0, S1PTW = 0 [ 0.664415] FSC = 0x04: level 0 translation fault [ 0.664421] Data abort info: [ 0.666050] ISV = 0, ISS = 0x00000004 [ 0.666399] CM = 0, WnR = 0 [ 0.666671] [0000000000000005] user address but active_mm is swapper [ 0.667240] Internal error: Oops: 96000004 [#1] PREEMPT SMP [ 0.667742] Modules linked in: [ 0.668028] CPU: 5 PID: 1 Comm: swapper/0 Not tainted 5.19.0-rc6 #300 [ 0.668608] Hardware name: Pine64 RockPro64 v2.0 (DT) [ 0.669062] pstate: 60000005 (nZCv daif -PAN -UAO -TCO -DIT -SSBS BTYPE=--) [ 0.669689] pc : pinctrl_lookup_state+0x20/0xc0 [ 0.670110] lr : rockchip_i2s_probe+0x1a8/0x54c [ 0.670527] sp : ffff80000a17bb30 [ 0.670829] x29: ffff80000a17bb30 x28: 0000000000000000 x27: ffff8000097c04c8 [ 0.671480] x26: ffff800009871060 x25: ffff800009871078 x24: ffff000001c11368 [ 0.672129] x23: ffff8000092dc850 x22: ffffffffffffffed x21: ffff8000096f7e98 [ 0.672776] x20: ffffffffffffffed x19: ffff000001d92480 x18: ffffffffffffffff [ 0.673423] x17: 000000040044ffff x16: ffff0000f77db2d0 x15: 0764076e07690766 [ 0.674070] x14: 0720076f07740720 x13: ffff800009e129f0 x12: 000000000000038d [ 0.674717] x11: 000000000000012f x10: ffff800009e6a9f0 x9 : ffff800009e129f0 [ 0.675364] x8 : 00000000ffffefff x7 : ffff800009e6a9f0 x6 : 80000000fffff000 [ 0.676011] x5 : 000000000000bff4 x4 : 0000000000000000 x3 : 0000000000000000 [ 0.676657] x2 : 0000000000000000 x1 : ffff8000096f7e98 x0 : ffffffffffffffed [ 0.677304] Call trace: [ 0.677531] pinctrl_lookup_state+0x20/0xc0 [ 0.677914] rockchip_i2s_probe+0x1a8/0x54c [ 0.678297] platform_probe+0x68/0xc0 [ 0.678638] really_probe.part.0+0x9c/0x2ac [ 0.679027] __driver_probe_device+0x98/0x144 [ 0.679429] driver_probe_device+0xac/0x140 [ 0.679814] __driver_attach+0xf8/0x184 [ 0.680169] bus_for_each_dev+0x70/0xd0 [ 0.680524] driver_attach+0x24/0x30 [ 0.680856] bus_add_driver+0x150/0x200 [ 0.681210] driver_register+0x78/0x130 [ 0.681560] __platform_driver_register+0x28/0x34 [ 0.681988] rockchip_i2s_driver_init+0x1c/0x28 [ 0.682407] do_one_initcall+0x50/0x1c0 [ 0.682760] kernel_init_freeable+0x204/0x288 [ 0.683160] kernel_init+0x28/0x13c [ 0.683482] ret_from_fork+0x10/0x20 [ 0.683816] Code: aa0003f4 a9025bf5 aa0003f6 aa0103f5 (f8418e93) [ 0.684365] ---[ end trace 0000000000000000 ]--- [ 0.684813] Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b [ 0.685500] SMP: stopping secondary CPUs [ 0.685995] Kernel Offset: disabled [ 0.686310] CPU features: 0x800,00105811,00001086 [ 0.686736] Memory Limit: none [ 0.687021] ---[ end Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b ]--- Check that i2s->pinctrl is valid before attempting to search for the bclk_on and bclk_off pinctrl states. Fixes: a5450aba737d ("ASoC: rockchip: i2s: switch BCLK to GPIO") Signed-off-by: Alexandru Elisei <alexandru.elisei(a)arm.com> --- Full log at [1], config at [2] (pastebins expire after 6 months). I'm not familiar with this part of the kernel, I did my best to come up with an explanation and a fix for the panic. Read Documentation/devicetree/bindings/sound/rockchip-i2s.yaml, which has the definition for the i2s nodes with the same compatible string as the i2s@ff8a0000 that is causing the panic (which is, "rockchip,rk3399-i2s" "rockchip,rk3066-i2s"). There's no mention there of a "pinctrl" property, maybe I'm reading the docs wrong, or maybe the board devicetree also needs fixing. [1] https://pastebin.com/vuRVDsKk [2] https://pastebin.com/3yDMF7YE sound/soc/rockchip/rockchip_i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 99a128a666fb..c9fedf6eb2e6 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -808,8 +808,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->bclk_ratio = 64; i2s->pinctrl = devm_pinctrl_get(&pdev->dev); - if (IS_ERR(i2s->pinctrl)) + if (IS_ERR(i2s->pinctrl)) { dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + ret = PTR_ERR(i2s->pinctrl); + goto err_clk; + } i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, "bclk_on"); -- 2.37.0

3 11

[PATCH v1 1/4] ALSA: hda: cs35l41: Improve dev_err_probe() messaging
by Andy Shevchenko 12 Jul '22

12 Jul '22

Drop duplicate print of returned value in the messages and use pattern return dev_err_probe(...) where it's possible. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/hda/cs35l41_hda.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index cce27a86267f..23a97ada7272 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -460,10 +460,8 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i dev_set_drvdata(dev, cs35l41); ret = cs35l41_hda_read_acpi(cs35l41, device_name, id); - if (ret) { - dev_err_probe(cs35l41->dev, ret, "Platform not supported %d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(cs35l41->dev, ret, "Platform not supported\n"); if (IS_ERR(cs35l41->reset_gpio)) { ret = PTR_ERR(cs35l41->reset_gpio); @@ -471,7 +469,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i if (ret == -EBUSY) { dev_info(cs35l41->dev, "Reset line busy, assuming shared reset\n"); } else { - dev_err_probe(cs35l41->dev, ret, "Failed to get reset GPIO: %d\n", ret); + dev_err_probe(cs35l41->dev, ret, "Failed to get reset GPIO\n"); goto err; } } -- 2.35.1

4 6

[PATCH] ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc671
by Meng Tang 12 Jul '22

12 Jul '22

On a HP 288 Pro G6, the front mic could not be detected.In order to get it working, the pin configuration needs to be set correctly, and the ALC671_FIXUP_HP_HEADSET_MIC2 fixup needs to be applied. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ecc737342fcc..34139c26795f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11219,6 +11219,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), + SND_PCI_QUIRK(0x103c, 0x877e, "HP 288 Pro G6", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), -- 2.20.1

2 1

[PATCH v1 1/1] ALSA: hda: cs35l41: Fix comments wrt serial-multi-instantiate reference
by Andy Shevchenko 12 Jul '22

12 Jul '22

The comments are inconsistent and point to the wrong driver name. The initially named i2c-multi-instantiate it was renamed to the serial-multi-instantiate exactly due to support of the platforms with multiple CS35L41 codecs. Fix comments accordingly. While at it, drop file names from the files. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/hda/cs35l41_hda.c | 4 ++-- sound/pci/hda/cs35l41_hda_i2c.c | 7 ++++--- sound/pci/hda/cs35l41_hda_spi.c | 7 ++++--- 3 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 23a97ada7272..49b25432a9f5 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -415,8 +415,8 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i no_acpi_dsd: /* * Device CLSA0100 doesn't have _DSD so a gpiod_get by the label reset won't work. - * And devices created by i2c-multi-instantiate don't have their device struct pointing to - * the correct fwnode, so acpi_dev must be used here. + * And devices created by serial-multi-instantiate don't have their device struct + * pointing to the correct fwnode, so acpi_dev must be used here. * And devm functions expect that the device requesting the resource has the correct * fwnode. */ diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index 5cfb5aac37ac..df39fc76e6be 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// cs35l41.c -- CS35l41 HDA I2C driver +// CS35l41 HDA I2C driver // // Copyright 2021 Cirrus Logic, Inc. // @@ -16,8 +16,9 @@ static int cs35l41_hda_i2c_probe(struct i2c_client *clt, const struct i2c_device { const char *device_name; - /* Compare against the device name so it works for I2C, normal ACPI - * and for ACPI by i2c-multi-instantiate matching cases + /* + * Compare against the device name so it works for SPI, normal ACPI + * and for ACPI by serial-multi-instantiate matching cases. */ if (strstr(dev_name(&clt->dev), "CLSA0100")) device_name = "CLSA0100"; diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index c9a61675c4fd..2f5afad3719e 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// cs35l41.c -- CS35l41 HDA SPI driver +// CS35l41 HDA SPI driver // // Copyright 2021 Cirrus Logic, Inc. // @@ -16,8 +16,9 @@ static int cs35l41_hda_spi_probe(struct spi_device *spi) { const char *device_name; - /* Compare against the device name so it works for SPI, normal ACPI - * and for ACPI by spi-multi-instantiate matching cases + /* + * Compare against the device name so it works for SPI, normal ACPI + * and for ACPI by serial-multi-instantiate matching cases. */ if (strstr(dev_name(&spi->dev), "CSC3551")) device_name = "CSC3551"; -- 2.35.1

2 1

[PATCH] ALSA: hda - Add fixup for Dell Latitidue E5430
by Meng Tang 12 Jul '22

12 Jul '22

Another Dell model, another fixup entry: Latitude E5430 needs the same fixup as other Latitude E series as workaround for noise problems. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fe7c1194d052..ecc737342fcc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8896,6 +8896,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05be, "Dell Latitude E6540", ALC292_FIXUP_DELL_E7X), -- 2.20.1

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[PATCH v2 0/2] dt-bindings: sound: Convert to json-schema
by Ryan.Wanner＠microchip.com 12 Jul '22

12 Jul '22

From: Ryan Wanner <Ryan.Wanner(a)microchip.com> This patch series converts atmel-classd and atmel-pdmic device tree bindings to json-schema. v1 -> v2: - Fix commit formatting. - Fix titles in yaml file - Removed trivial descriptions - Correct formatting errors Ryan Wanner (2): dt-bindings: sound: atmel,classd: Convert to json-schema dt-binding: sound: atmel,pdmic: Convert to json-schema .../bindings/sound/atmel,sama5d2-classd.yaml | 104 ++++++++++++++++++ .../bindings/sound/atmel,sama5d2-pdmic.yaml | 98 +++++++++++++++++ .../bindings/sound/atmel-classd.txt | 55 --------- .../devicetree/bindings/sound/atmel-pdmic.txt | 55 --------- 4 files changed, 202 insertions(+), 110 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt delete mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt -- 2.34.1

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axfer: fix typo in manual
by GitHub pull_request - opened 12 Jul '22

12 Jul '22

alsa-project/alsa-utils pull request #156 was opened from takaswie: The spelling of 'aborted' was 'aboeted' in the manual. This commit fixes it. Fixes: a37703614a90 ("axfer: fulfill manual section for libffado backend") Reported-by: Chao Song chao.song(a)linux.intel.com Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-utils/pull/156 Patch URL : https://github.com/alsa-project/alsa-utils/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 00/12] ASoC: Intel: avs: Updates and cleanups
by Cezary Rojewski 11 Jul '22

11 Jul '22

Series consists of loosely connected patches and does not concentrate on one specific subject. First, as generic HDAudio codec driver is now part of ASoC, avs-driver core is updated to register missing ext_bus operations. This completes driver's core implementation. The next change adds the last missing piece for port descriptions coming from topology in formatted string format e.g.: ssp%d have full effect. To do that, the port value needs to be provided to respective copier configuration. Third change relaxes core transition timings so that scenarios where modules are interfering with each other while being on separate cores are not occasionally causing trouble. All other changes are addressing warnings, cleaning things up a little and protecting driver from invalid firmware behavior - while not expected in release binaries, does not hurt to add them. Amadeusz Sławiński (2): ASoC: Intel: avs: Set max DMA segment size ASoC: Intel: avs: Use helper function to set up DMA Cezary Rojewski (10): ASoC: Intel: avs: Register HDAudio ext-bus operations ASoC: Intel: avs: Assign I2S gateway when parsing topology ASoC: Intel: avs: Relax DSP core transition timings ASoC: Intel: avs: Copy only as many RX bytes as necessary ASoC: Intel: avs: Shield LARGE_CONFIG_GETs against zero payload_size ASoC: Intel: avs: Block IPC channel on suspend ASoC: Intel: avs: Recognize FW_CFG_RESERVED ASoC: Intel: avs: Replace hardcodes with SD_CTL_STREAM_RESET ASoC: Intel: avs: Lower UNLOAD_MULTIPLE_MODULES IPC timeout ASoC: Intel: avs: Update AVS_FW_INIT_TIMEOUT_US declaration sound/soc/intel/Kconfig | 2 +- sound/soc/intel/avs/cldma.c | 12 ++++++------ sound/soc/intel/avs/core.c | 13 ++++++------- sound/soc/intel/avs/dsp.c | 11 +++++++++-- sound/soc/intel/avs/ipc.c | 1 + sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 18 +++++++++++++----- sound/soc/intel/avs/topology.c | 27 +++++++++++++++++++++++++++ 8 files changed, 64 insertions(+), 22 deletions(-) -- 2.25.1

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[PATCH] ASoC: nau8821: Add headset button detection
by Seven Lee 11 Jul '22

11 Jul '22

This patch adds the function of headphone button detection, Button detection will be enabled if the device tree has a key_enable property. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 2 +- sound/soc/codecs/nau8821.c | 35 +++++++++++++++++++ sound/soc/codecs/nau8821.h | 1 + 3 files changed, 37 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt index 6c3baf7a5f21..7c84e7c7327a 100644 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ b/Documentation/devicetree/bindings/sound/nau8821.txt @@ -34,7 +34,7 @@ Optional properties: - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - + - nuvoton,key_enable: Headset button detection switch. Example: diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index ce4e7f46bb06..7b0a9c8db40a 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -29,6 +29,8 @@ #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 +#define NAU8821_BUTTON SND_JACK_BTN_0 + /* the maximum frequency of CLK_ADC and CLK_DAC */ #define CLK_DA_AD_MAX 6144000 @@ -911,6 +913,20 @@ static void nau8821_eject_jack(struct nau8821 *nau8821) /* Recover to normal channel input */ regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, NAU8821_ADC_R_SRC_EN, 0); + if (nau8821->key_enable) { + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN); + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS); + } + } static void nau8821_jdet_work(struct work_struct *work) @@ -940,6 +956,15 @@ static void nau8821_jdet_work(struct work_struct *work) */ regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, NAU8821_ADC_R_SRC_EN, NAU8821_ADC_R_SRC_EN); + if (nau8821->key_enable) { + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN, 0); + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS, 0); + } } else { dev_dbg(nau8821->dev, "Headphone connected\n"); event |= SND_JACK_HEADPHONE; @@ -999,6 +1024,13 @@ static irqreturn_t nau8821_interrupt(int irq, void *data) nau8821_eject_jack(nau8821); event_mask |= SND_JACK_HEADSET; clear_irq = NAU8821_JACK_EJECT_IRQ_MASK; + } else if (active_irq & NAU8821_KEY_SHORT_PRESS_IRQ) { + event |= NAU8821_BUTTON; + event_mask |= NAU8821_BUTTON; + clear_irq = NAU8821_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8821_KEY_RELEASE_IRQ) { + event_mask = NAU8821_BUTTON; + clear_irq = NAU8821_KEY_RELEASE_IRQ; } else if ((active_irq & NAU8821_JACK_INSERT_IRQ_MASK) == NAU8821_JACK_INSERT_DETECTED) { regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, @@ -1490,6 +1522,7 @@ static void nau8821_print_device_properties(struct nau8821 *nau8821) nau8821->jack_eject_debounce); dev_dbg(dev, "dmic-clk-threshold: %d\n", nau8821->dmic_clk_threshold); + dev_dbg(dev, "key_enable: %d\n", nau8821->key_enable); } static int nau8821_read_device_properties(struct device *dev, @@ -1503,6 +1536,8 @@ static int nau8821_read_device_properties(struct device *dev, "nuvoton,jkdet-pull-enable"); nau8821->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); + nau8821->key_enable = device_property_read_bool(dev, + "nuvoton,key-enable"); ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8821->jkdet_polarity); if (ret) diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index a92edfeb9d3a..c44251f54d48 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -525,6 +525,7 @@ struct nau8821 { int jack_eject_debounce; int fs; int dmic_clk_threshold; + int key_enable; }; int nau8821_enable_jack_detect(struct snd_soc_component *component, -- 2.25.1

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