September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3 1/2] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 01 Oct '19

01 Oct '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 262 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------- 2 files changed, 262 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..faa75b91c072 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,262 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the + second being the connection's source. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.21.0

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[alsa-devel] [PATCH v2 00/14] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 01 Oct '19

01 Oct '19

Hi Mark These are v2 of soc-core.c cleanup (maybe) step 3. These try to clenaup around kmalloc/kfree related code. Basically these do nothing from technical point of view. I added +1 extra patch as [00/14] Kuninori Morimoto (14): ASoC: soc-core: move soc_free_pcm_runtime() ASoC: soc-core: merge soc_add_pcm_runtime() into soc_new_pcm_runtime() ASoC: soc-core: call list_del(&rtd->list) at soc_free_pcm_runtime() ASoC: soc-core: create rtd->codec_dais first ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init() ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order ASoC: soc-core: remove snd_soc_rtdcom_del_all() ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() ASoC: soc-ops: use snd_soc_card_get_kcontrol() at snd_soc_limit_volume() include/sound/soc.h | 1 - sound/soc/soc-core.c | 287 ++++++++++++++++++++++++++------------------------- sound/soc/soc-ops.c | 11 +- 3 files changed, 151 insertions(+), 148 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v7] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 01 Oct '19

01 Oct '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 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. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

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[alsa-devel] [PATCH] ASoC: stm32: sai: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index ef4273361d0d..e20267504b16 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -100,7 +100,7 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - stm32_sai_pclk_disable(&sai->pdev->dev); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } -- 2.20.1

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[alsa-devel] [PATCH] ASoC: wcd9335: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is an if statement that is indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wcd9335.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f318403133e9..f11ffa28683b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2837,11 +2837,11 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, TX_HPF_CUT_OFF_FREQ_MASK) >> 5; snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x10); snd_soc_component_update_bits(comp, dec_cfg_reg, 0x08, 0x00); - if (hpf_coff_freq != CF_MIN_3DB_150HZ) { - snd_soc_component_update_bits(comp, dec_cfg_reg, - TX_HPF_CUT_OFF_FREQ_MASK, - hpf_coff_freq << 5); - } + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + hpf_coff_freq << 5); + } break; case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rt5663: clean up indentation issues
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There are two break statements that are indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5663.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..e6c1ec6c426e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3644,7 +3644,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, RT5663_LDO1_DVO_MASK | RT5663_AMP_HP_MASK, RT5663_LDO1_DVO_0_9V | RT5663_AMP_HP_3X); - break; + break; case CODEC_VER_0: regmap_update_bits(rt5663->regmap, RT5663_DIG_MISC, RT5663_DIG_GATE_CTRL_MASK, RT5663_DIG_GATE_CTRL_EN); @@ -3663,7 +3663,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_MASK, RT5663_DATA_SWAP_ADCDAT1_LL); - break; + break; default: dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); } -- 2.20.1

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[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: core: Clarify usage of ignore_machine
by Daniel Baluta 01 Oct '19

01 Oct '19

For a sound card ignore_machine means that existing FEs links should be ignored and existing BEs links should be overridden with some information from the matching component driver. Current code make some confusions about this so fix it! Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 88978a3036c4..e32a45f6bd88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1853,7 +1853,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) for_each_component(component) { - /* does this component override FEs ? */ + /* does this component override BEs ? */ if (!component->driver->ignore_machine) continue; @@ -1874,7 +1874,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; } - dev_info(card->dev, "info: override FE DAI link %s\n", + dev_info(card->dev, "info: override BE DAI link %s\n", card->dai_link[i].name); /* override platform component */ -- 2.17.1

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[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 01 Oct '19

01 Oct '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' Changes since rfc: - Fixed missing error assignments (copy pasted the error) - Document the trigger order in the commit message Regards, Peter sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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