diff options
37 files changed, 376 insertions, 165 deletions
diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml index d51431df7acf..b5594a9d508e 100644 --- a/Documentation/devicetree/bindings/sound/everest,es8326.yaml +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -24,6 +24,10 @@ properties: items: - const: mclk + interrupts: + maxItems: 1 + description: interrupt output for headset detection + "#sound-dai-cells": const: 0 diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml index d1b4e23f1c95..27c34ce4c2e2 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -18,11 +18,15 @@ description: properties: compatible: - enum: - - fsl,imx35-esai - - fsl,imx6ull-esai - - fsl,imx8qm-esai - - fsl,vf610-esai + oneOf: + - enum: + - fsl,imx35-esai + - fsl,imx6ull-esai + - fsl,vf610-esai + - items: + - enum: + - fsl,imx8qm-esai + - const: fsl,imx6ull-esai reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 204f361cea27..5654e9f61aba 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -16,16 +16,23 @@ description: | properties: compatible: - enum: - - fsl,imx35-spdif - - fsl,vf610-spdif - - fsl,imx6sx-spdif - - fsl,imx8qm-spdif - - fsl,imx8qxp-spdif - - fsl,imx8mq-spdif - - fsl,imx8mm-spdif - - fsl,imx8mn-spdif - - fsl,imx8ulp-spdif + oneOf: + - items: + - enum: + - fsl,imx35-spdif + - fsl,imx6sx-spdif + - fsl,imx8mm-spdif + - fsl,imx8mn-spdif + - fsl,imx8mq-spdif + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + - fsl,imx8ulp-spdif + - fsl,vf610-spdif + - items: + - enum: + - fsl,imx6sl-spdif + - fsl,imx6sx-spdif + - const: fsl,imx35-spdif reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml index f94ad0715e32..ba482747f0e6 100644 --- a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml @@ -29,6 +29,13 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8188 ASoC platform. + mediatek,adsp: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the MT8188 ADSP platform, which is the optional Audio DSP + hardware that provides additional audio functionalities if present. + The AFE will link to ADSP when the phandle is provided. + patternProperties: "^dai-link-[0-9]+$": type: object diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 2e2e01493a5f..b9e33a7429b0 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -25,6 +25,7 @@ properties: - enum: - qcom,sm8550-sndcard - qcom,sm8650-sndcard + - qcom,sm8750-sndcard - const: qcom,sm8450-sndcard - enum: - qcom,apq8096-sndcard diff --git a/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml b/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml new file mode 100644 index 000000000000..c15c01bbb884 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd,pcm-platform.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sprd,pcm-platform.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Spreadtrum DMA platform + +maintainers: + - Orson Zhai <orsonzhai@gmail.com> + - Baolin Wang <baolin.wang7@gmail.com> + - Chunyan Zhang <zhang.lyra@gmail.com> + +properties: + compatible: + const: sprd,pcm-platform + + dmas: + maxItems: 10 + + dma-names: + items: + - const: normal_p_l + - const: normal_p_r + - const: normal_c_l + - const: normal_c_r + - const: voice_c + - const: fast_p + - const: loop_c + - const: loop_p + - const: voip_c + - const: voip_p + +required: + - compatible + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + platform { + compatible = "sprd,pcm-platform"; + dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, + <&agcp_dma 3 3>, <&agcp_dma 4 4>, + <&agcp_dma 5 5>, <&agcp_dma 6 6>, + <&agcp_dma 7 7>, <&agcp_dma 8 8>, + <&agcp_dma 9 9>, <&agcp_dma 10 10>; + dma-names = "normal_p_l", "normal_p_r", + "normal_c_l", "normal_c_r", + "voice_c", "fast_p", + "loop_c", "loop_p", + "voip_c", "voip_p"; + }; +... diff --git a/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml b/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml new file mode 100644 index 000000000000..3b66bedeff97 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sprd,sc9860-mcdt.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sprd,sc9860-mcdt.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Spreadtrum Multi-Channel Data Transfer controller + +description: + The Multi-channel data transfer controller is used for sound stream + transmission between the audio subsystem and other AP/CP subsystem. It + supports 10 DAC channels and 10 ADC channels, and each channel can be + configured with DMA mode or interrupt mode. + +maintainers: + - Orson Zhai <orsonzhai@gmail.com> + - Baolin Wang <baolin.wang7@gmail.com> + - Chunyan Zhang <zhang.lyra@gmail.com> + +properties: + compatible: + const: sprd,sc9860-mcdt + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + mcdt@41490000 { + compatible = "sprd,sc9860-mcdt"; + reg = <0x41490000 0x170>; + interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/sprd-mcdt.txt b/Documentation/devicetree/bindings/sound/sprd-mcdt.txt deleted file mode 100644 index 274ba0acbfd6..000000000000 --- a/Documentation/devicetree/bindings/sound/sprd-mcdt.txt +++ /dev/null @@ -1,19 +0,0 @@ -Spreadtrum Multi-Channel Data Transfer Binding - -The Multi-channel data transfer controller is used for sound stream -transmission between audio subsystem and other AP/CP subsystem. It -supports 10 DAC channel and 10 ADC channel, and each channel can be -configured with DMA mode or interrupt mode. - -Required properties: -- compatible: Should be "sprd,sc9860-mcdt". -- reg: Should contain registers address and length. -- interrupts: Should contain one interrupt shared by all channel. - -Example: - -mcdt@41490000 { - compatible = "sprd,sc9860-mcdt"; - reg = <0 0x41490000 0 0x170>; - interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; -}; diff --git a/Documentation/devicetree/bindings/sound/sprd-pcm.txt b/Documentation/devicetree/bindings/sound/sprd-pcm.txt deleted file mode 100644 index fbbcade2181d..000000000000 --- a/Documentation/devicetree/bindings/sound/sprd-pcm.txt +++ /dev/null @@ -1,23 +0,0 @@ -* Spreadtrum DMA platform bindings - -Required properties: -- compatible: Should be "sprd,pcm-platform". -- dmas: Specify the list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: Identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. - -Example: - - audio_platform:platform@0 { - compatible = "sprd,pcm-platform"; - dmas = <&agcp_dma 1 1>, <&agcp_dma 2 2>, - <&agcp_dma 3 3>, <&agcp_dma 4 4>, - <&agcp_dma 5 5>, <&agcp_dma 6 6>, - <&agcp_dma 7 7>, <&agcp_dma 8 8>, - <&agcp_dma 9 9>, <&agcp_dma 10 10>; - dma-names = "normal_p_l", "normal_p_r", - "normal_c_l", "normal_c_r", - "voice_c", "fast_p", - "loop_c", "loop_p", - "voip_c", "voip_p"; - }; diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0d1b215f24f4..b275201b02f6 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -216,8 +216,7 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int rollback); void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); -int snd_soc_dai_compress_new(struct snd_soc_dai *dai, - struct snd_soc_pcm_runtime *rtd, int num); +int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd); bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int stream); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); @@ -275,7 +274,7 @@ struct snd_soc_dai_ops { int (*probe)(struct snd_soc_dai *dai); int (*remove)(struct snd_soc_dai *dai); /* compress dai */ - int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); + int (*compress_new)(struct snd_soc_pcm_runtime *rtd); /* Optional Callback used at pcm creation*/ int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/include/sound/soc.h b/include/sound/soc.h index 5c240ea34027..4f5d411e3823 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -486,11 +486,11 @@ struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name); -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_SND_SOC_COMPRESS -int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd); #else -static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +static inline int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd) { return 0; } @@ -1195,7 +1195,7 @@ struct snd_soc_pcm_runtime { struct dentry *debugfs_dpcm_root; #endif - unsigned int num; /* 0-based and monotonic increasing */ + unsigned int id; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ /* function mark */ diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index a25f94d6eb67..0e82598e10af 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -152,14 +152,15 @@ void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_lin struct snd_soc_dai_link_component *cpus, int cpus_num, struct snd_soc_dai_link_component *platform_component, int num_platforms, struct snd_soc_dai_link_component *codecs, - int codecs_num, int (*init)(struct snd_soc_pcm_runtime *rtd), + int codecs_num, int no_pcm, + int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops); int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, int *be_id, char *name, int playback, int capture, const char *cpu_dai_name, const char *platform_comp_name, int num_platforms, const char *codec_name, - const char *codec_dai_name, + const char *codec_dai_name, int no_pcm, int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops); diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 67aa0ad83486..d314253207d5 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1561,7 +1561,6 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) if (drv_data->dmic_cpu_id == DMIC) { links[i].name = "acp-dmic-codec"; - links[i].stream_name = "DMIC capture"; links[i].id = DMIC_BE_ID; links[i].codecs = dmic_codec; links[i].num_codecs = ARRAY_SIZE(dmic_codec); diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 36e6d6db90c1..8fce8cb957c9 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -236,7 +236,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, cpus, num_cpus, platform_component, ARRAY_SIZE(platform_component), codecs, num_codecs, - asoc_sdw_rtd_init, &sdw_ops); + 1, asoc_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations @@ -285,7 +285,7 @@ static int create_sdw_dailinks(struct snd_soc_card *card, } static int create_dmic_dailinks(struct snd_soc_card *card, - struct snd_soc_dai_link **dai_links, int *be_id) + struct snd_soc_dai_link **dai_links, int *be_id, int no_pcm) { struct device *dev = card->dev; int ret; @@ -294,7 +294,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, 0, 1, // DMIC only supports capture "acp-sof-dmic", platform_component->name, ARRAY_SIZE(platform_component), - "dmic-codec", "dmic-hifi", + "dmic-codec", "dmic-hifi", no_pcm, asoc_sdw_dmic_init, NULL); if (ret) return ret; @@ -377,7 +377,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (ctx->ignore_internal_dmic) { dev_warn(dev, "Ignoring ACP DMIC\n"); } else { - ret = create_dmic_dailinks(card, &dai_links, &be_id); + ret = create_dmic_dailinks(card, &dai_links, &be_id, 1); if (ret) return ret; } diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index 018f2372e892..e3a4fcc63a56 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -256,12 +256,16 @@ static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv) offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >> I2S_RX_DESC_OFF_LEVEL_SHIFT; + bool val_read = false; while (offlevel) { regmap_read(regmap_i2s, I2S_RX_DESC_OFF_ADDR, &val_1); regmap_read(regmap_i2s, I2S_RX_DESC_OFF_LEN, &val_2); + val_read = true; offlevel--; } - prtd->dma_addr_next = val_1 + val_2; + if (val_read) + prtd->dma_addr_next = val_1 + val_2; + ifflevel = (int_status & I2S_RX_DESC_IFF_LEVEL_MASK) >> I2S_RX_DESC_IFF_LEVEL_SHIFT; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 311ea7918b31..e2da3e317b5a 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1167,17 +1167,20 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); int ret = 0; - if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) + mutex_lock(&da7219->pll_lock); + + if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) { + mutex_unlock(&da7219->pll_lock); return 0; + } if ((freq < 2000000) || (freq > 54000000)) { + mutex_unlock(&da7219->pll_lock); dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", freq); return -EINVAL; } - mutex_lock(&da7219->pll_lock); - switch (clk_id) { case DA7219_CLKSRC_MCLK_SQR: snd_soc_component_update_bits(component, DA7219_PLL_CTRL, diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index aa3e364827c8..a5603b617688 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x00); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); } } } else { @@ -1082,7 +1082,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x03); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c index c0f8cccae3b2..c71453da088a 100644 --- a/sound/soc/codecs/rt721-sdca-sdw.c +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -203,7 +203,7 @@ static int rt721_sdca_update_status(struct sdw_slave *slave, * This also could sync with the cache value as the rt721_sdca_jack_init set. */ sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); } @@ -280,7 +280,7 @@ static int rt721_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 900; + prop->clk_stop_timeout = 1380; /* wake-up event */ prop->wake_capable = 1; @@ -337,11 +337,6 @@ static int rt721_sdca_interrupt_callback(struct sdw_slave *slave, SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); if (ret < 0) goto io_error; - } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { - ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, - SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); - if (ret < 0) - goto io_error; } ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); if (ret < 0) @@ -475,7 +470,7 @@ static int __maybe_unused rt721_sdca_dev_system_suspend(struct device *dev) mutex_lock(&rt721_sdca->disable_irq_lock); rt721_sdca->disable_irq = true; ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + SDW_SCP_SDCA_INTMASK_SDCA_0, 0); ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8, 0); mutex_unlock(&rt721_sdca->disable_irq_lock); @@ -502,7 +497,7 @@ static int __maybe_unused rt721_sdca_dev_resume(struct device *dev) if (!slave->unattach_request) { mutex_lock(&rt721->disable_irq_lock); if (rt721->disable_irq == true) { - sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt721->disable_irq = false; } diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index bdd160b80b64..1c9f32e405cf 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -39,7 +39,7 @@ static void rt721_sdca_jack_detect_handler(struct work_struct *work) return; /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ - if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { rt721->jack_type = rt_sdca_headset_detect(rt721->regmap, RT721_SDCA_ENT_GE49); if (rt721->jack_type < 0) @@ -286,7 +286,7 @@ static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) mutex_lock(&rt721->calibrate_mutex); if (rt721->hs_jack) { sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); dev_dbg(&rt721->slave->dev, "in %s enable\n", __func__); @@ -298,6 +298,8 @@ static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) regmap_write(rt721->regmap, SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU0D, RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_XU_REL_CTRL, 0x0000); rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, RT721_GE_REL_CTRL1, 0x4000, 0x4000); } diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h index e2f071909da8..0a82c107b19a 100644 --- a/sound/soc/codecs/rt721-sdca.h +++ b/sound/soc/codecs/rt721-sdca.h @@ -133,6 +133,7 @@ struct rt721_sdca_dmic_kctrl_priv { #define RT721_HDA_LEGACY_UAJ_CTL 0x02 #define RT721_HDA_LEGACY_CTL1 0x05 #define RT721_HDA_LEGACY_RESET_CTL 0x06 +#define RT721_XU_REL_CTRL 0x0c #define RT721_GE_REL_CTRL1 0x0d #define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e #define RT721_GE_SDCA_RST_CTRL 0x10 diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 87354bb1564e..0abbd92cbc7e 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -177,7 +177,7 @@ static int rt722_sdca_update_status(struct sdw_slave *slave, * This also could sync with the cache value as the rt722_sdca_jack_init set. */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); } @@ -308,12 +308,8 @@ static int rt722_sdca_interrupt_callback(struct sdw_slave *slave, SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); if (ret < 0) goto io_error; - } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { - ret = sdw_update_no_pm(rt722->slave, SDW_SCP_SDCA_INT1, - SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); - if (ret < 0) - goto io_error; } + ret = sdw_read_no_pm(rt722->slave, SDW_SCP_SDCA_INT2); if (ret < 0) goto io_error; @@ -444,7 +440,7 @@ static int __maybe_unused rt722_sdca_dev_system_suspend(struct device *dev) mutex_lock(&rt722_sdca->disable_irq_lock); rt722_sdca->disable_irq = true; ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + SDW_SCP_SDCA_INTMASK_SDCA_0, 0); ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8, 0); mutex_unlock(&rt722_sdca->disable_irq_lock); @@ -471,7 +467,7 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) if (!slave->unattach_request) { mutex_lock(&rt722->disable_irq_lock); if (rt722->disable_irq == true) { - sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt722->disable_irq = false; } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index f9f7512ca360..908846e994df 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -190,8 +190,8 @@ static void rt722_sdca_jack_detect_handler(struct work_struct *work) if (!rt722->component->card || !rt722->component->card->instantiated) return; - /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ - if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + /* SDW_SCP_SDCA_INT_SDCA_0 is used for jack detection */ + if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { ret = rt722_sdca_headset_detect(rt722); if (ret < 0) return; @@ -294,7 +294,7 @@ static void rt722_sdca_jack_init(struct rt722_sdca_priv *rt722) if (rt722->hs_jack) { /* set SCP_SDCA_IntMask1[0]=1 */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); /* set SCP_SDCA_IntMask2[0]=1 */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); @@ -308,6 +308,7 @@ static void rt722_sdca_jack_init(struct rt722_sdca_priv *rt722) regmap_write(rt722->regmap, SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT722_SDCA_ENT_XU0D, RT722_SDCA_CTL_SELECTED_MODE, 0), 0); + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_GE_RELATED_CTL1, 0x0000); /* trigger GE interrupt */ rt722_sdca_index_update_bits(rt722, RT722_VENDOR_HDA_CTL, RT722_GE_RELATED_CTL2, 0x4000, 0x4000); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e283751abfef..8e88830e8e57 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -30,6 +30,7 @@ config SND_SOC_FSL_MQS tristate "Medium Quality Sound (MQS) module support" depends on SND_SOC_FSL_SAI select REGMAP_MMIO + select IMX_SCMI_MISC_DRV if IMX_SCMI_MISC_EXT !=n help Say Y if you want to add Medium Quality Sound (MQS) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index 145f9ca15e43..0513e9e8402e 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -6,6 +6,7 @@ // Copyright 2019 NXP #include <linux/clk.h> +#include <linux/firmware/imx/sm.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/mfd/syscon.h> @@ -74,6 +75,29 @@ struct fsl_mqs { #define FSL_MQS_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define FSL_MQS_FORMATS SNDRV_PCM_FMTBIT_S16_LE +static int fsl_mqs_sm_read(void *context, unsigned int reg, unsigned int *val) +{ + struct fsl_mqs *mqs_priv = context; + int num = 1; + + if (IS_ENABLED(CONFIG_IMX_SCMI_MISC_DRV) && + mqs_priv->soc->ctrl_off == reg) + return scmi_imx_misc_ctrl_get(SCMI_IMX_CTRL_MQS1_SETTINGS, &num, val); + + return -EINVAL; +}; + +static int fsl_mqs_sm_write(void *context, unsigned int reg, unsigned int val) +{ + struct fsl_mqs *mqs_priv = context; + + if (IS_ENABLED(CONFIG_IMX_SCMI_MISC_DRV) && + mqs_priv->soc->ctrl_off == reg) + return scmi_imx_misc_ctrl_set(SCMI_IMX_CTRL_MQS1_SETTINGS, val); + + return -EINVAL; +}; + static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -188,6 +212,13 @@ static const struct regmap_config fsl_mqs_regmap_config = { .cache_type = REGCACHE_NONE, }; +static const struct regmap_config fsl_mqs_sm_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_read = fsl_mqs_sm_read, + .reg_write = fsl_mqs_sm_write, +}; + static int fsl_mqs_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; @@ -219,6 +250,16 @@ static int fsl_mqs_probe(struct platform_device *pdev) dev_err(&pdev->dev, "failed to get gpr regmap\n"); return PTR_ERR(mqs_priv->regmap); } + } else if (mqs_priv->soc->type == TYPE_REG_SM) { + mqs_priv->regmap = devm_regmap_init(&pdev->dev, + NULL, + mqs_priv, + &fsl_mqs_sm_regmap); + if (IS_ERR(mqs_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(mqs_priv->regmap)); + return PTR_ERR(mqs_priv->regmap); + } } else { regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 0f11f20dc51a..95a57fda0250 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -275,7 +275,7 @@ static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct imx_card_data *data = snd_soc_card_get_drvdata(rtd->card); const struct imx_card_plat_data *plat_data = data->plat_data; - struct dai_link_data *link_data = &data->link_data[rtd->num]; + struct dai_link_data *link_data = &data->link_data[rtd->id]; unsigned int width = slots * slot_width; unsigned int rate = params_rate(params); int i; @@ -313,7 +313,7 @@ static int imx_aif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_card *card = rtd->card; struct imx_card_data *data = snd_soc_card_get_drvdata(card); - struct dai_link_data *link_data = &data->link_data[rtd->num]; + struct dai_link_data *link_data = &data->link_data[rtd->id]; struct imx_card_plat_data *plat_data = data->plat_data; struct device *dev = card->dev; struct snd_soc_dai *codec_dai; @@ -435,7 +435,7 @@ static int imx_aif_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct imx_card_data *data = snd_soc_card_get_drvdata(card); - struct dai_link_data *link_data = &data->link_data[rtd->num]; + struct dai_link_data *link_data = &data->link_data[rtd->id]; static struct snd_pcm_hw_constraint_list constraint_rates; static struct snd_pcm_hw_constraint_list constraint_channels; int ret = 0; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index fedae7f6f70c..d47c372228b3 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -296,7 +296,7 @@ int simple_util_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); struct simple_util_dai *dai; unsigned int fixed_sysclk = 0; int i1, i2, i; @@ -357,7 +357,7 @@ void simple_util_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); struct simple_util_dai *dai; int i; @@ -448,7 +448,7 @@ int simple_util_hw_params(struct snd_pcm_substream *substream, struct simple_util_dai *pdai; struct snd_soc_dai *sdai; struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); unsigned int mclk, mclk_fs = 0; int i, ret; @@ -517,7 +517,7 @@ int simple_util_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->id); struct simple_util_data *data = &dai_props->adata; struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); @@ -628,7 +628,7 @@ static int simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, int simple_util_dai_init(struct snd_soc_pcm_runtime *rtd) { struct simple_util_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *props = simple_priv_to_props(priv, rtd->id); struct simple_util_dai *dai; int i, ret; diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5614e706a0bb..eaba91dad967 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -484,10 +484,26 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF6") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF9") }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CFA") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, /* MeteorLake devices */ { .callback = sof_sdw_quirk_cb, @@ -576,6 +592,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0D36") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF8") }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), @@ -647,6 +671,30 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF3") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF4") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF5") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, /* Pantherlake devices*/ { .callback = sof_sdw_quirk_cb, @@ -790,7 +838,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, cpus, num_cpus, platform_component, ARRAY_SIZE(platform_component), codecs, num_codecs, - asoc_sdw_rtd_init, &sdw_ops); + 1, asoc_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations @@ -867,7 +915,7 @@ static int create_ssp_dailinks(struct snd_soc_card *card, playback, capture, cpu_dai_name, platform_component->name, ARRAY_SIZE(platform_component), codec_name, - ssp_info->dais[0].dai_name, NULL, + ssp_info->dais[0].dai_name, 1, NULL, ssp_info->ops); if (ret) return ret; @@ -892,7 +940,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, 0, 1, // DMIC only supports capture "DMIC01 Pin", platform_component->name, ARRAY_SIZE(platform_component), - "dmic-codec", "dmic-hifi", + "dmic-codec", "dmic-hifi", 1, asoc_sdw_dmic_init, NULL); if (ret) return ret; @@ -903,7 +951,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, 0, 1, // DMIC only supports capture "DMIC16k Pin", platform_component->name, ARRAY_SIZE(platform_component), - "dmic-codec", "dmic-hifi", + "dmic-codec", "dmic-hifi", 1, /* don't call asoc_sdw_dmic_init() twice */ NULL, NULL); if (ret) @@ -947,7 +995,7 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, 1, 0, // HDMI only supports playback cpu_dai_name, platform_component->name, ARRAY_SIZE(platform_component), - codec_name, codec_dai_name, + codec_name, codec_dai_name, 1, i == 0 ? sof_sdw_hdmi_init : NULL, NULL); if (ret) return ret; @@ -975,7 +1023,7 @@ static int create_bt_dailinks(struct snd_soc_card *card, 1, 1, cpu_dai_name, platform_component->name, ARRAY_SIZE(platform_component), snd_soc_dummy_dlc.name, snd_soc_dummy_dlc.dai_name, - NULL, NULL); + 1, NULL, NULL); if (ret) return ret; diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 5ebf287fe700..a2dfccb7990f 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -43,7 +43,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = - (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->id]; return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); } @@ -56,7 +56,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) { struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = - (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->id]; struct snd_soc_dai *codec_dai; int ret, i; @@ -86,7 +86,7 @@ static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) { struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = - (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->id]; int ret; /* The loopback rx_mask is the pad tx_mask */ diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 455f6bfc9f8f..b408cc2bbc91 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -32,7 +32,7 @@ static int gx_card_i2s_be_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct gx_dai_link_i2s_data *be = - (struct gx_dai_link_i2s_data *)priv->link_data[rtd->num]; + (struct gx_dai_link_i2s_data *)priv->link_data[rtd->id]; return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); } diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 922ecada1cd8..311377317176 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -190,6 +190,7 @@ static const struct of_device_id snd_sc8280xp_dt_match[] = { {.compatible = "qcom,sm8450-sndcard", "sm8450"}, {.compatible = "qcom,sm8550-sndcard", "sm8550"}, {.compatible = "qcom,sm8650-sndcard", "sm8650"}, + {.compatible = "qcom,sm8750-sndcard", "sm8750"}, {} }; diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; diff --git a/sound/soc/renesas/rcar/core.c b/sound/soc/renesas/rcar/core.c index c32e88d6a141..e2234928c9e8 100644 --- a/sound/soc/renesas/rcar/core.c +++ b/sound/soc/renesas/rcar/core.c @@ -1843,7 +1843,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = name, .info = rsnd_kctrl_info, - .index = rtd->num, + .index = rtd->id, .get = rsnd_kctrl_get, .put = rsnd_kctrl_put, }; diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 6610efe8af18..19bd02e2cd6d 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1015,15 +1015,17 @@ void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_lin struct snd_soc_dai_link_component *cpus, int cpus_num, struct snd_soc_dai_link_component *platform_component, int num_platforms, struct snd_soc_dai_link_component *codecs, - int codecs_num, int (*init)(struct snd_soc_pcm_runtime *rtd), + int codecs_num, int no_pcm, + int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops) { dev_dbg(dev, "create dai link %s, id %d\n", name, *be_id); dai_links->id = (*be_id)++; dai_links->name = name; + dai_links->stream_name = name; dai_links->platforms = platform_component; dai_links->num_platforms = num_platforms; - dai_links->no_pcm = 1; + dai_links->no_pcm = no_pcm; dai_links->cpus = cpus; dai_links->num_cpus = cpus_num; dai_links->codecs = codecs; @@ -1039,7 +1041,7 @@ int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *d int *be_id, char *name, int playback, int capture, const char *cpu_dai_name, const char *platform_comp_name, int num_platforms, const char *codec_name, - const char *codec_dai_name, + const char *codec_dai_name, int no_pcm, int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops) { @@ -1058,7 +1060,7 @@ int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *d asoc_sdw_init_dai_link(dev, dai_links, be_id, name, playback, capture, &dlc[0], 1, &dlc[1], num_platforms, - &dlc[2], 1, init, ops); + &dlc[2], 1, no_pcm, init, ops); return 0; } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index a0c55246f424..3c514703fa33 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -537,11 +537,10 @@ static struct snd_compr_ops soc_compr_dyn_ops = { * snd_soc_new_compress - create a new compress. * * @rtd: The runtime for which we will create compress - * @num: the device index number (zero based - shared with normal PCMs) * * Return: 0 for success, else error. */ -int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); @@ -617,7 +616,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) snprintf(new_name, sizeof(new_name), "(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, rtd->id, playback, capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, @@ -638,7 +637,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else { snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, codec_dai->name, rtd->id); memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); } @@ -652,7 +651,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) break; } - ret = snd_compress_new(rtd->card->snd_card, num, direction, + ret = snd_compress_new(rtd->card->snd_card, rtd->id, direction, new_name, compr); if (ret < 0) { component = snd_soc_rtd_to_codec(rtd, 0)->component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f04b671ce33e..a1dace4bb616 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -558,7 +558,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( */ rtd->card = card; rtd->dai_link = dai_link; - rtd->num = card->num_rtd++; + rtd->id = card->num_rtd++; rtd->pmdown_time = pmdown_time; /* default power off timeout */ /* see for_each_card_rtds */ @@ -1166,7 +1166,7 @@ static int snd_soc_add_pcm_runtime(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link_component *codec, *platform, *cpu; struct snd_soc_component *component; - int i, ret; + int i, id, ret; lockdep_assert_held(&client_mutex); @@ -1225,6 +1225,28 @@ static int snd_soc_add_pcm_runtime(struct snd_soc_card *card, } } + /* + * Most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + * + * FIXME + * + * This should be implemented by using "dai_link" feature instead of + * "component" feature. + */ + id = rtd->id; + for_each_rtd_components(rtd, i, component) { + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + id += component->driver->be_pcm_base; + else + id = rtd->dai_link->id; + } + rtd->id = id; + return 0; _err_defer: @@ -1457,8 +1479,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); - struct snd_soc_component *component; - int ret, num, i; + int ret; /* do machine specific initialization */ ret = snd_soc_link_init(rtd); @@ -1473,30 +1494,13 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); - num = rtd->num; - - /* - * most drivers will register their PCMs using DAI link ordering but - * topology based drivers can use the DAI link id field to set PCM - * device number and then use rtd + a base offset of the BEs. - */ - for_each_rtd_components(rtd, i, component) { - if (!component->driver->use_dai_pcm_id) - continue; - - if (rtd->dai_link->no_pcm) - num += component->driver->be_pcm_base; - else - num = rtd->dai_link->id; - } - /* create compress_device if possible */ - ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); + ret = snd_soc_dai_compress_new(cpu_dai, rtd); if (ret != -ENOTSUPP) goto err; /* create the pcm */ - ret = soc_new_pcm(rtd, num); + ret = soc_new_pcm(rtd); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 4a1c85ad5a8d..34ba1a93a4c9 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -457,12 +457,12 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, } int snd_soc_dai_compress_new(struct snd_soc_dai *dai, - struct snd_soc_pcm_runtime *rtd, int num) + struct snd_soc_pcm_runtime *rtd) { int ret = -ENOTSUPP; if (dai->driver->ops && dai->driver->ops->compress_new) - ret = dai->driver->ops->compress_new(rtd, num); + ret = dai->driver->ops->compress_new(rtd); return soc_dai_ret(dai, ret); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 678400e76e53..fb7f25fd8ec5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2891,7 +2891,7 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, static int soc_create_pcm(struct snd_pcm **pcm, struct snd_soc_pcm_runtime *rtd, - int playback, int capture, int num) + int playback, int capture) { char new_name[64]; int ret; @@ -2901,13 +2901,13 @@ static int soc_create_pcm(struct snd_pcm **pcm, snprintf(new_name, sizeof(new_name), "codec2codec(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, rtd->id, playback, capture, pcm); } else if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", rtd->dai_link->stream_name); - ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, rtd->id, playback, capture, pcm); } else { if (rtd->dai_link->dynamic) @@ -2916,9 +2916,9 @@ static int soc_create_pcm(struct snd_pcm **pcm, else snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, - soc_codec_dai_name(rtd), num); + soc_codec_dai_name(rtd), rtd->id); - ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, + ret = snd_pcm_new(rtd->card->snd_card, new_name, rtd->id, playback, capture, pcm); } if (ret < 0) { @@ -2926,13 +2926,13 @@ static int soc_create_pcm(struct snd_pcm **pcm, new_name, rtd->dai_link->name, ret); return ret; } - dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n",num, new_name); + dev_dbg(rtd->card->dev, "ASoC: registered pcm #%d %s\n", rtd->id, new_name); return 0; } /* create a new pcm */ -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; struct snd_pcm *pcm; @@ -2943,7 +2943,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (ret < 0) return ret; - ret = soc_create_pcm(&pcm, rtd, playback, capture, num); + ret = soc_create_pcm(&pcm, rtd, playback, capture); if (ret < 0) return ret; |