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

[PATCH v2] ALSA: hda/realtek: Add quirk for another Asus K42JZ model
by Meng Tang 05 Aug '22

05 Aug '22

There is another Asus K42JZ model with the PCI SSID 1043:1313 that requires the quirk ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE. Add the corresponding entry to the quirk table. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 383a814b8539..4461237a3474 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6842,6 +6842,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, + ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -7427,6 +7428,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK, }, + [ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a110f0 }, /* use as headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -9124,6 +9134,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for another Asus K42JZ model
by Meng Tang 05 Aug '22

05 Aug '22

There is another Asus K42JZ model with the PCI SSID 1043:1313 that requires the quirk ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE. Add the corresponding entry to the quirk table. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 383a814b8539..4461237a3474 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6842,6 +6842,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, + ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -7427,6 +7428,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK, }, + [ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a110f0 }, /* use as headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + } [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -9124,6 +9134,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), -- 2.20.1

1 0

[PATCH] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 05 Aug '22

05 Aug '22

In commit <c3ecef21c3f26> ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ffc24afb5a7a..f0602077b385 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1054,7 +1054,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

1 0

[PATCH v2] ASoC: amd: acp: Add TDM support for acp i2s stream
by Venkata Prasad Potturu 05 Aug '22

05 Aug '22

Add callback and code changes to enable ACP I2S controller in TDM mode. Add new fields in acp_stream and acp_dev_data struct to configure tdm related registers for ACP i2s controllers. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Depends on:- -- https://patchwork.kernel.org/project/alsa-devel/patch/20220804072556.601396… -- https://patchwork.kernel.org/project/alsa-devel/patch/20220801063501.51439-… Changes since v1: -- Add spin lock where linked list is referring. sound/soc/amd/acp/acp-i2s.c | 80 ++++++++++++++++++++++++++++++++++++- sound/soc/amd/acp/amd.h | 12 ++++++ 2 files changed, 91 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 393f729ef561..ac416572db0d 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -25,6 +25,65 @@ #define DRV_NAME "acp_i2s_playcap" +static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + int mode; + + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (mode) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = TDM_DISABLE; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = TDM_ENABLE; + break; + default: + return -EINVAL; + } + return 0; +} + +static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mask, + int slots, int slot_width) +{ + struct device *dev = dai->component->dev; + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); + struct acp_stream *stream; + int slot_len; + + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + dev_err(dev, "Unsupported bitdepth %d\n", slot_width); + return -EINVAL; + } + + spin_lock_irq(&adata->acp_lock); + list_for_each_entry(stream, &adata->stream_list, list) { + if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + adata->tdm_tx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) + adata->tdm_rx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + } + spin_unlock_irq(&adata->acp_lock); + return 0; +} + static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -33,7 +92,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ struct acp_resource *rsrc; u32 val; u32 xfer_resolution; - u32 reg_val; + u32 reg_val, fmt_reg, tdm_fmt; u32 lrclk_div_val, bclk_div_val; adata = snd_soc_dai_get_drvdata(dai); @@ -62,12 +121,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_ITER; + fmt_reg = ACP_BTTDM_TXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_ITER; + fmt_reg = ACP_I2STDM_TXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_ITER; + fmt_reg = ACP_HSTDM_TXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -77,12 +139,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_IRER; + fmt_reg = ACP_BTTDM_RXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_IRER; + fmt_reg = ACP_I2STDM_RXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_IRER; + fmt_reg = ACP_HSTDM_RXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -95,6 +160,16 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ val = val | (xfer_resolution << 3); writel(val, adata->acp_base + reg_val); + if (adata->tdm_mode) { + val = readl(adata->acp_base + reg_val); + writel(val | BIT(1), adata->acp_base + reg_val); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tdm_fmt = adata->tdm_tx_fmt[dai->driver->id - 1]; + else + tdm_fmt = adata->tdm_rx_fmt[dai->driver->id - 1]; + writel(tdm_fmt, adata->acp_base + fmt_reg); + } + if (rsrc->soc_mclk) { switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -443,6 +518,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d stream->id = dai->driver->id + dir; stream->dai_id = dai->driver->id; stream->irq_bit = irq_bit; + stream->dir = substream->stream; return 0; } @@ -452,6 +528,8 @@ const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops = { .hw_params = acp_i2s_hwparams, .prepare = acp_i2s_prepare, .trigger = acp_i2s_trigger, + .set_fmt = acp_i2s_set_fmt, + .set_tdm_slot = acp_i2s_set_tdm_slot, }; EXPORT_SYMBOL_NS_GPL(asoc_acp_cpu_dai_ops, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index be8bb8247c4e..5f2119f42271 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -84,6 +84,14 @@ #define ACP_MAX_STREAM 8 +#define TDM_ENABLE 1 +#define TDM_DISABLE 0 + +#define SLOT_WIDTH_8 0x8 +#define SLOT_WIDTH_16 0x10 +#define SLOT_WIDTH_24 0x18 +#define SLOT_WIDTH_32 0x20 + struct acp_chip_info { char *name; /* Platform name */ unsigned int acp_rev; /* ACP Revision id */ @@ -96,6 +104,7 @@ struct acp_stream { int irq_bit; int dai_id; int id; + int dir; u64 bytescount; u32 reg_offset; u32 pte_offset; @@ -120,6 +129,7 @@ struct acp_dev_data { void __iomem *acp_base; unsigned int i2s_irq; + bool tdm_mode; /* SOC specific dais */ struct snd_soc_dai_driver *dai_driver; int num_dai; @@ -134,6 +144,8 @@ struct acp_dev_data { u32 lrclk_div; struct acp_resource *rsrc; + u32 tdm_tx_fmt[3]; + u32 tdm_rx_fmt[3]; }; union acp_i2stdm_mstrclkgen { -- 2.25.1

1 0

[PATCH] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 05 Aug '22

05 Aug '22

In commit <c3ecef21c3f26> ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ffc24afb5a7a..f0602077b385 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1054,7 +1054,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

2 2

[PATCH] ASoC: fsl_asrc_dma: fully initialize structs
by Sascha Hauer 05 Aug '22

05 Aug '22

The driver uses two statically ininitialized struct dma_slave_config, but only one of them is initialized to zero. Initialize config_be to zero as well to make sure that no fields are filled with random values. Let the compiler do this instead of explicitly calling memset() which makes it easier to read. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_asrc_dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cba..fec56dbb64e60 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -138,7 +138,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct dma_chan *tmp_chan = NULL, *be_chan = NULL; struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; - struct dma_slave_config config_fe, config_be; + struct dma_slave_config config_fe = {}, config_be = {}; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -181,7 +181,6 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, return -EINVAL; } - memset(&config_fe, 0, sizeof(config_fe)); ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); if (ret) { dev_err(dev, "failed to prepare DMA config for Front-End\n"); -- 2.30.2

3 3

[PATCH 0/2] ASoC: SOF: Introduce of_machine_select
by Chunxu Li 04 Aug '22

04 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> In these patches, we introduce of_machine_select for SOF Chunxu Li (2): ASoC: SOF: Introduce optional callback of_machine_select ASoC: SOF: mediatek: Add .of_machine_select field for mt8186 include/sound/sof.h | 2 ++ sound/soc/sof/mediatek/mt8186/mt8186.c | 11 +++++++++++ sound/soc/sof/ops.h | 9 +++++++++ sound/soc/sof/pcm.c | 8 +++++++- sound/soc/sof/sof-audio.c | 7 +++++++ sound/soc/sof/sof-of-dev.c | 23 +++++++++++++++++++++++ sound/soc/sof/sof-of-dev.h | 8 ++++++++ sound/soc/sof/sof-priv.h | 1 + 8 files changed, 68 insertions(+), 1 deletion(-) -- 2.25.1

3 9

[PATCH v2] dt-bindings: sound: tlv320aic3x: Convert to dtschema
by Jai Luthra 04 Aug '22

04 Aug '22

Convert bindings for TI's TLV320AIC3x audio codecs to dtschema. Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- .../bindings/sound/ti,tlv320aic3x.yaml | 158 ++++++++++++++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 97 ----------- 2 files changed, 158 insertions(+), 97 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml new file mode 100644 index 000000000000..f0ed856162ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -0,0 +1,158 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic3x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC3x Codec + +description: | + TLV320AIC3x are a series of low-power stereo audio codecs with stereo + headphone amplifier, as well as multiple inputs and outputs programmable in + single-ended or fully differential configurations. + + The serial control bus supports SPI or I2C protocols, while the serial audio + data bus is programmable for I2S, left/right-justified, DSP, or TDM modes. + + The following pins can be referred in the sound node's audio routing property: + + CODEC output pins: + LLOUT + RLOUT + MONO_LOUT + HPLOUT + HPROUT + HPLCOM + HPRCOM + + CODEC input pins for TLV320AIC3104: + MIC2L + MIC2R + LINE1L + LINE1R + + CODEC input pins for other compatible codecs: + MIC3L + MIC3R + LINE1L + LINE2L + LINE1R + LINE2R + +maintainers: + - Jai Luthra <j-luthra(a)ti.com> + +properties: + compatible: + enum: + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + gpio-reset: + maxItems: 1 + description: + Deprecated, please use reset-gpios instead. + deprecated: true + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + ai3x-ocmv = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 20931a63fd64..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,97 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality - - Not supported on tlv320aic3104 -- ai3x-micbias-vg - MicBias Voltage required. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the - device as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * LLOUT - * RLOUT - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -I2C example: - -#include <dt-bindings/gpio/gpio.h> - -tlv320aic3x: tlv320aic3x@1b { - compatible = "ti,tlv320aic3x"; - reg = <0x1b>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; - -SPI example: - -spi0: spi@f0000000 { - tlv320aic3x: codec@0 { - compatible = "ti,tlv320aic3x"; - reg = <0>; /* CS number */ - #sound-dai-cells = <0>; - spi-max-frequency = <1000000>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; - ai3x-ocmv = <0>; - }; -}; -- 2.17.1

3 2

[PATCH v2 0/3] DT binding for sample format conversion
by Sameer Pujar 04 Aug '22

04 Aug '22

DT binding properties are available to fixup rate and channel parameters of a DAI. This series extends this to sample format conversion as well. With this now DAI PCM parameters (channels, sample rate and sample format) can be fixed up as necessary in an audio path. Changelog: ========== v1->v2: ------- * Move DAI params properties to a new schema and re-use this for simple-card and audio-graph-card. * Use string type for DAI format binding as suggested by Krzysztof. Sameer Pujar (3): ASoC: dt-bindings: Add schema for common DAI params ASoC: dt-bindings: Add sample format conversion ASoC: simple-card-utils: Fixup DAI sample format .../bindings/sound/audio-graph-port.yaml | 17 +++++----- .../devicetree/bindings/sound/audio-graph.yaml | 9 ++--- .../devicetree/bindings/sound/dai-params.yaml | 38 ++++++++++++++++++++++ .../devicetree/bindings/sound/simple-card.yaml | 20 +++++------- include/sound/simple_card_utils.h | 1 + sound/soc/generic/simple-card-utils.c | 34 +++++++++++++++++++ 6 files changed, 95 insertions(+), 24 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/dai-params.yaml -- 2.7.4

2 9

[PATCH v4] ASoC: amd: acp: Initialize list to store acp_stream during pcm_open
by Venkata Prasad Potturu 04 Aug '22

04 Aug '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 v3: -- Modify to spin_lock() in irq handler and spin_lock_irq() in dma open and close functions instead of spin_lock_irqsave(). Changes since v2: -- Add locking mechanism in irq handler, linked list updatation and deletion. Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-platform.c | 40 +++++++++++++++----------------- sound/soc/amd/acp/amd.h | 4 +++- 2 files changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index f561d39b33e2..beee53aedeaf 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; @@ -104,14 +104,13 @@ 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]; + spin_lock(&adata->acp_lock); + 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)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } if (adata->rsrc->no_of_ctrls == 2) { if (stream && (ext_intr_stat1 & stream->irq_bit)) { @@ -119,10 +118,10 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } } } + spin_unlock(&adata->acp_lock); if (i2s_flag) return IRQ_HANDLED; @@ -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 +183,10 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs return -ENOMEM; stream->substream = substream; - adata->stream[stream_id] = stream; + + spin_lock_irq(&adata->acp_lock); + list_add_tail(&stream->list, &adata->stream_list); + spin_unlock_irq(&adata->acp_lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp_pcm_hardware_playback; @@ -212,16 +210,13 @@ 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 +256,15 @@ 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 */ + spin_lock_irq(&adata->acp_lock); + list_del(&stream->list); + spin_unlock_irq(&adata->acp_lock); kfree(stream); - adata->stream[stream_id] = NULL; return 0; } @@ -305,6 +299,10 @@ 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); + spin_lock_init(&adata->acp_lock); + 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 60a7c95f134f..be8bb8247c4e 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 +124,8 @@ 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; + spinlock_t acp_lock; struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; -- 2.25.1

1 0