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

Applied "dt-bindings: sound: Add TLV320ADCx140 dt bindings" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch dt-bindings: sound: Add TLV320ADCx140 dt bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4ee67cbd97668ab1b17d86d85348302c0b7490cd Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Thu, 20 Feb 2020 15:07:58 -0600 Subject: [PATCH] dt-bindings: sound: Add TLV320ADCx140 dt bindings Add dt bindings for the TLV320ADCx140 Burr-Brown ADC. The initial support is for the following: TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200220210759.31466-2-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/tlv320adcx140.yaml | 83 +++++++++++++++++++ 1 file changed, 83 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml new file mode 100644 index 000000000000..1433ff62b14f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320adcx140.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital + PDM microphones recording), high-performance audio, analog-to-digital + converter (ADC) with analog inputs supporting up to 2V RMS. The TLV320ADCX140 + family supports line and microphone Inputs, and offers a programmable + microphone bias or supply voltage generation. + + Specifications can be found at: + http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf + http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf + http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf + +properties: + compatible: + oneOf: + - const: ti,tlv320adc3140 + - const: ti,tlv320adc5140 + - const: ti,tlv320adc6140 + + reg: + maxItems: 1 + description: | + I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpios: + description: | + GPIO used for hardware reset. + + areg-supply: + description: | + Regulator with AVDD at 3.3V. If not defined then the internal regulator + is enabled. + + ti,mic-bias-source: + description: | + Indicates the source for MIC Bias. + 0 - Mic bias is set to VREF + 1 - Mic bias is set to VREF × 1.096 + 6 - Mic bias is set to AVDD + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + - enum: [0, 1, 6] + + ti,vref-source: + description: | + Indicates the source for MIC Bias. + 0 - Set VREF to 2.75V + 1 - Set VREF to 2.5V + 2 - Set VREF to 1.375V + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + - enum: [0, 1, 2] + +required: + - compatible + - reg + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tlv320adc5140"; + reg = <0x4c>; + ti,use-internal-areg; + ti,mic-bias-source = <6>; + reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; + }; + }; -- 2.20.1

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[PATCH for-next v3 0/2] Introduce the TLV320ADCx140 codec family
by Dan Murphy 21 Feb '20

21 Feb '20

Hello Introducing the Texas Instruments TLV320ADCx140 quad channel audio ADC. This device supports 4 analog inputs, 8 digital inputs or a combination of analog and digital microphones. TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Dan Murphy (2): dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family .../bindings/sound/tlv320adcx140.yaml | 83 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adcx140.c | 849 ++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 130 +++ 5 files changed, 1073 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h -- 2.25.0

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[PATCH v3] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 20 Feb '20

20 Feb '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - use pattern for compatible of child nodes - rework dmas and clocks properties - add "additionalProperties" Changes in v3: - move clocks properties for st,stm32h7-sai compatible, to 'else' clause --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 193 ++++++++++++++++++ 2 files changed, 193 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index 944743dd9212..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..cdea20150c1e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,193 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + + additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + items: + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: x8k + - const: x11k + + - else: + properties: + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: x8k + - const: x11k + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai1: sai@4400a000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400a000 0x400>; + reg = <0x4400a000 0x4>, <0x4400a3f0 0x10>; + interrupts = <GIC_SPI 87 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&rcc SAI1>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "x8k", "x11k"; + resets = <&rcc SAI1_R>; + + sai1a: audio-controller@4400a004 { + compatible = "st,stm32-sai-sub-a"; + #sound-dai-cells = <0>; + reg = <0x4 0x1c>; + clocks = <&rcc SAI1_K>; + clock-names = "sai_ck"; + dmas = <&dmamux1 87 0x400 0x01>; + dma-names = "tx"; + }; + }; + +... -- 2.17.1

2 1

[PATCH] ASoC: tas2562: Add support for digital volume control
by Dan Murphy 20 Feb '20

20 Feb '20

Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 94 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 98 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b2682c2360b6..e577aa6f304a 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -334,6 +353,22 @@ static int tas2562_codec_probe(struct snd_soc_component *component) if (ret < 0) return ret; + /* Set the Digital volume to -110dB */ + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, 0x00); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, 0x00); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, 0x0d); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, 0x43); + if (ret) + return ret; + + tas2562->volume_lvl = 0; + return 0; } @@ -414,6 +449,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -427,6 +506,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -517,6 +607,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -- 2.25.0

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[PATCH] sound: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 20 Feb '20

20 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertently introduced[3] to the codebase from now on. Also, notice that, dynamic memory allocations won't be affected by this change: "Flexible array members have incomplete type, and so the sizeof operator may not be applied. As a quirk of the original implementation of zero-length arrays, sizeof evaluates to zero."[1] This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- include/sound/control.h | 2 +- include/sound/intel-nhlt.h | 6 +++--- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/asound.h | 2 +- include/uapi/sound/skl-tplg-interface.h | 2 +- sound/core/oss/pcm_plugin.h | 2 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/soc/intel/atom/sst-atom-controls.h | 2 +- sound/soc/intel/skylake/skl-i2s.h | 2 +- sound/soc/intel/skylake/skl-topology.h | 4 ++-- sound/soc/intel/skylake/skl.h | 2 +- sound/usb/usx2y/usbusx2y.h | 2 +- 13 files changed, 16 insertions(+), 16 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 11feeee31e35..aeaed2a05bae 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -75,7 +75,7 @@ struct snd_kcontrol { unsigned long private_value; void *private_data; void (*private_free)(struct snd_kcontrol *kcontrol); - struct snd_kcontrol_volatile vd[0]; /* volatile data */ + struct snd_kcontrol_volatile vd[]; /* volatile data */ }; #define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list) diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index f657fd8fc0ad..743c2f442280 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -50,7 +50,7 @@ enum nhlt_device_type { struct nhlt_specific_cfg { u32 size; - u8 caps[0]; + u8 caps[]; } __packed; struct nhlt_fmt_cfg { @@ -60,7 +60,7 @@ struct nhlt_fmt_cfg { struct nhlt_fmt { u8 fmt_count; - struct nhlt_fmt_cfg fmt_config[0]; + struct nhlt_fmt_cfg fmt_config[]; } __packed; struct nhlt_endpoint { @@ -80,7 +80,7 @@ struct nhlt_endpoint { struct nhlt_acpi_table { struct acpi_table_header header; u8 endpoint_count; - struct nhlt_endpoint desc[0]; + struct nhlt_endpoint desc[]; } __packed; struct nhlt_resource_desc { diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1a9c5dd40228..f870f927b70c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -688,7 +688,7 @@ struct snd_soc_dapm_context { /* A list of widgets associated with an object, typically a snd_kcontrol */ struct snd_soc_dapm_widget_list { int num_widgets; - struct snd_soc_dapm_widget *widgets[0]; + struct snd_soc_dapm_widget *widgets[]; }; #define for_each_dapm_widgets(list, i, widget) \ diff --git a/include/sound/soc.h b/include/sound/soc.h index f0e4f36f83bf..9944b978ae70 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1146,7 +1146,7 @@ struct snd_soc_pcm_runtime { unsigned int fe_compr:1; /* for Dynamic PCM */ int num_components; - struct snd_soc_component *components[0]; /* CPU/Codec/Platform */ + struct snd_soc_component *components[]; /* CPU/Codec/Platform */ }; #define for_each_rtd_components(rtd, i, component) \ for ((i) = 0; \ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..3809993f6cb4 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -1071,7 +1071,7 @@ struct snd_ctl_elem_value { struct snd_ctl_tlv { unsigned int numid; /* control element numeric identification */ unsigned int length; /* in bytes aligned to 4 */ - unsigned int tlv[0]; /* first TLV */ + unsigned int tlv[]; /* first TLV */ }; #define SNDRV_CTL_IOCTL_PVERSION _IOR('U', 0x00, int) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index 9eee32f5e407..62f76f154d19 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -149,7 +149,7 @@ struct skl_dfw_algo_data { __u32 rsvd:30; __u32 param_id; __u32 max; - char params[0]; + char params[]; } __packed; enum skl_tkn_dir { diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 8d2f7a4e3ab6..46e273bd4a78 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -64,7 +64,7 @@ struct snd_pcm_plugin { char *buf; snd_pcm_uframes_t buf_frames; struct snd_pcm_plugin_channel *buf_channels; - char extra_data[0]; + char extra_data[]; }; int snd_pcm_plugin_build(struct snd_pcm_substream *handle, diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index dda797209a27..654e28a6669f 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -177,7 +177,7 @@ struct snd_efw_phys_meters { u32 in_meters; u32 reserved4; u32 reserved5; - u32 values[0]; + u32 values[]; } __packed; enum snd_efw_clock_source { SND_EFW_CLOCK_SOURCE_INTERNAL = 0, diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index 5356e954a732..620b48d2a064 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -410,7 +410,7 @@ struct sst_cmd_set_gain_dual { struct sst_cmd_set_params { struct sst_destination_id dst; u16 command_id; - char params[0]; + char params[]; } __packed; diff --git a/sound/soc/intel/skylake/skl-i2s.h b/sound/soc/intel/skylake/skl-i2s.h index d7c15873c0d4..dfce91e11be1 100644 --- a/sound/soc/intel/skylake/skl-i2s.h +++ b/sound/soc/intel/skylake/skl-i2s.h @@ -46,7 +46,7 @@ struct skl_i2s_config_mclk { struct skl_i2s_config_mclk_ext { u32 mdivctrl; u32 mdivr_count; - u32 mdivr[0]; + u32 mdivr[]; } __packed; struct skl_i2s_config_blob_signature { diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index e967800dbb62..d2cd8ef8e97f 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -119,7 +119,7 @@ struct skl_cpr_gtw_cfg { struct skl_dma_control { u32 node_id; u32 config_length; - u32 config_data[0]; + u32 config_data[]; } __packed; struct skl_cpr_cfg { @@ -152,7 +152,7 @@ struct skl_up_down_mixer_cfg { struct skl_algo_cfg { struct skl_base_cfg base_cfg; - char params[0]; + char params[]; } __packed; struct skl_base_outfmt_cfg { diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 2bfbf59277c4..26057f38a014 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -49,7 +49,7 @@ struct skl_astate_param { struct skl_astate_config { u32 count; - struct skl_astate_param astate_table[0]; + struct skl_astate_param astate_table[]; }; struct skl_fw_config { diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index e0f77172ce8f..144b85f57bd2 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -18,7 +18,7 @@ struct snd_usX2Y_AsyncSeq { struct snd_usX2Y_urbSeq { int submitted; int len; - struct urb *urb[0]; + struct urb *urb[]; }; #include "usx2yhwdeppcm.h" -- 2.25.0

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[PATCH v2] ASoC: tas2562: Add support for digital volume control
by Dan Murphy 20 Feb '20

20 Feb '20

Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Fix init values to lower volume CFG4 is the LSB and CFG1 is MSB sound/soc/codecs/tas2562.c | 94 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 98 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b2682c2360b6..92e4bf34b35c 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -334,6 +353,22 @@ static int tas2562_codec_probe(struct snd_soc_component *component) if (ret < 0) return ret; + /* Set the Digital volume to -110dB */ + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, 0x43); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, 0x0d); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, 0x00); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, 0x00); + if (ret) + return ret; + + tas2562->volume_lvl = 0; + return 0; } @@ -414,6 +449,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -427,6 +506,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -517,6 +607,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -- 2.25.0

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Re: [alsa-devel] No sound since 5.4 on skl_n88l25_s4567
by Kuninori Morimoto 20 Feb '20

20 Feb '20

Hi ojab > > Please take a look, I want my sound back ._. > > Forgot to mention, it's reproducible on linux master git. Hmm... This patch 1) removes unneeded if check 2) adds missing error check. 1) shouldn't have effect to your PC. Thus, maybe your PC failed by 2). It had been worked by accident before ? I guess your kernel is telling some message like this ? ASoC: no dapm match for %s --> %s --> %s Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH] ASoC: samsung: Update dependencies for Arizona machine drivers
by Charles Keepax 20 Feb '20

20 Feb '20

Currently it is possible to get the following bad config: WARNING: unmet direct dependencies detected for SND_SOC_WM5110 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n] commit ea00d95200d0 ("ASoC: Use imply for SND_SOC_ALL_CODECS") commit d8dd3f92a6ba ("ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout") After these two patches the machine drivers still selects the SND_SOC_WM5110 symbol which doesn't take account of the dependency added on the MFD_WM5110 symbol, fix this by also adding a dependency on MFD_WM5110 itself. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1a0b163ca47b5..112911dc271ba 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -151,7 +151,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on MFD_ARIZONA && I2C && SPI_MASTER + depends on MFD_ARIZONA && MFD_WM5102 && MFD_WM5110 && I2C && SPI_MASTER depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 @@ -204,7 +204,7 @@ config SND_SOC_ARNDALE config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" - depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && MFD_WM5110 && I2C && SPI_MASTER depends on GPIOLIB || COMPILE_TEST select SND_SOC_MAX98504 select SND_SOC_WM5110 -- 2.11.0

2 1

[PATCH] ASoC: soc-core: fix for_rtd_codec_dai_rollback() macro
by Pierre-Louis Bossart 20 Feb '20

20 Feb '20

The use of parentheses to protect the argument is fine for (i)++ but not for (--i). Fixes: 83f94a2e293d61 ("ASoC: soc-core: add snd_soc_close_delayed_work()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f0e4f36f83bf..8a2266676b2d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1157,7 +1157,7 @@ struct snd_soc_pcm_runtime { ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ (i)++) #define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ - for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);) + for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); -- 2.20.1

3 2

[PATCH] ASoC: dpcm: remove confusing trace in dpcm_get_be()
by Jerome Brunet 20 Feb '20

20 Feb '20

Now that dpcm_get_be() is used in dpcm_end_walk_at_be(), it is not a error if this function does not find a BE for the provided widget. Remove the related dev_err() trace which is confusing since things might be working as expected. When called from dpcm_add_paths(), it is an error if dpcm_get_be() fails to find a BE for the provided widget. The necessary error trace is already done in this case. Fixes: 027a48387183 ("ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..aff27c8599ef 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1270,9 +1270,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, } } - /* dai link name and stream name set correctly ? */ - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", - stream ? "capture" : "playback", widget->name); + /* Widget provided is not a BE */ return NULL; } -- 2.24.1

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