June 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: hdmi-codec: select CONFIG_HDMI
by Arnd Bergmann 02 Jun '16

02 Jun '16

SND_SOC_HDMI_CODEC can be enabled without HDMI support, leading to a link error: In function `hdmi_codec_hw_params': sound/soc/codecs/hdmi-codec.c:188: undefined reference to `hdmi_audio_infoframe_init' sound/built-in.o:(.debug_addr+0x1a5c0): undefined reference to `hdmi_audio_infoframe_init' This changes the Kconfig file to select HDMI, as the other codec using hdmi_audio_infoframe_init already does. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 09184118a8ab ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") --- sound/soc/codecs/Kconfig | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4d82a58ff6b0..f3fb98f0a995 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -483,9 +483,10 @@ config SND_SOC_DMIC tristate config SND_SOC_HDMI_CODEC - tristate - select SND_PCM_ELD - select SND_PCM_IEC958 + tristate + select SND_PCM_ELD + select SND_PCM_IEC958 + select HDMI config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" -- 2.7.0

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[alsa-devel] [PATCH 1/2] ASoC: Add ADAU7002 Stereo PDM-to-I2S/TDM Converter DT bindings
by Lars-Peter Clausen 02 Jun '16

02 Jun '16

The ADAU7002 takes a stereo PDM signal (e.g. from two digital microphones) and converts it into a I2S/TDM PCM stream. The chip does not have a control interface and has a single power supply that is modeled in the devicetree. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/adi,adau7002.txt | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt new file mode 100644 index 0000000..f144ee1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.txt @@ -0,0 +1,19 @@ +Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +Required properties: + + - compatible: Must be "adi,adau7002" + +Optional properties: + + - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD + supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt + + If this property is not present it is assumed that the supply pin is + hardwired to always on. + +Example: + adau7002: pdm-to-i2s { + compatible = "adi,adau7002"; + IOVDD-supply = <&supply>; + }; -- 2.1.4

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix dra7 DMA offset when using CFG port
by Peter Ujfalusi 02 Jun '16

02 Jun '16

The TX and RX offset is different for each serializers when using the CFG port for DMA access. When using the CFG port only one serializer can be used per direction so print error message and only configure the first serializer's offset. Reported-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Suggested-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++++++--- sound/soc/davinci/davinci-mcasp.h | 4 +-- 2 files changed, 54 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e060798d94cb..f8cb3c7d3c45 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1534,8 +1534,9 @@ static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { - .tx_dma_offset = 0x200, - .rx_dma_offset = 0x284, + /* The CFG port offset will be calculated if it is needed */ + .tx_dma_offset = 0, + .rx_dma_offset = 0, .version = MCASP_VERSION_4, }; @@ -1755,6 +1756,52 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) return PCM_EDMA; } +static u32 davinci_mcasp_txdma_offset(struct davinci_mcasp_pdata *pdata) +{ + int i; + u32 offset = 0; + + if (pdata->version != MCASP_VERSION_4) + return pdata->tx_dma_offset; + + for (i = 0; i < pdata->num_serializer; i++) { + if (pdata->serial_dir[i] == TX_MODE) { + if (!offset) { + offset = DAVINCI_MCASP_TXBUF_REG(i); + } else { + pr_err("%s: Only one serializer allowed!\n", + __func__); + break; + } + } + } + + return offset; +} + +static u32 davinci_mcasp_rxdma_offset(struct davinci_mcasp_pdata *pdata) +{ + int i; + u32 offset = 0; + + if (pdata->version != MCASP_VERSION_4) + return pdata->rx_dma_offset; + + for (i = 0; i < pdata->num_serializer; i++) { + if (pdata->serial_dir[i] == RX_MODE) { + if (!offset) { + offset = DAVINCI_MCASP_RXBUF_REG(i); + } else { + pr_err("%s: Only one serializer allowed!\n", + __func__); + break; + } + } + } + + return offset; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1883,7 +1930,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) dma_data->addr = dat->start; else - dma_data->addr = mem->start + pdata->tx_dma_offset; + dma_data->addr = mem->start + davinci_mcasp_txdma_offset(pdata); dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -1904,7 +1951,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) dma_data->addr = dat->start; else - dma_data->addr = mem->start + pdata->rx_dma_offset; + dma_data->addr = + mem->start + davinci_mcasp_rxdma_offset(pdata); dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a3be108a8c17..10b52d6bdeca 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -85,9 +85,9 @@ (n << 2)) /* Transmit Buffer for Serializer n */ -#define DAVINCI_MCASP_TXBUF_REG 0x200 +#define DAVINCI_MCASP_TXBUF_REG(n) (0x200 + (n << 2)) /* Receive Buffer for Serializer n */ -#define DAVINCI_MCASP_RXBUF_REG 0x280 +#define DAVINCI_MCASP_RXBUF_REG(n) (0x280 + (n << 2)) /* McASP FIFO Registers */ #define DAVINCI_MCASP_V2_AFIFO_BASE (0x1010) -- 2.8.3

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Use a copy of pdata per instance during DT boot
by Peter Ujfalusi 02 Jun '16

02 Jun '16

Instead of modifying the static pdata struct per McASP instance we need to allocate pdata for each McASP. This way we can avoid configuration leakage from prior McASP to McASP drivers probed at later time. Reported-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f8cb3c7d3c45..e10cfda21d89 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1620,7 +1620,14 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( pdata = pdev->dev.platform_data; return pdata; } else if (match) { - pdata = (struct davinci_mcasp_pdata*) match->data; + pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata), + GFP_KERNEL); + if (!pdata) { + dev_err(&pdev->dev, + "Failed to allocate memory for pdata\n"); + ret = -ENOMEM; + return pdata; + } } else { /* control shouldn't reach here. something is wrong */ ret = -EINVAL; -- 2.8.3

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[alsa-devel] [PATCH] ASoC: sgtl5000: Place optional properties in the correct section
by Fabio Estevam 02 Jun '16

02 Jun '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> 'micbias-resistor-k-ohms' and 'micbias-voltage-m-volts' are optional properties, so move them below the 'Optional properties' line. While at it, fix a typo in 'mentioned'. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 0e5e4eb..5666da7 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -7,6 +7,14 @@ Required properties: - clocks : the clock provider of SYS_MCLK +- VDDA-supply : the regulator provider of VDDA + +- VDDIO-supply: the regulator provider of VDDIO + +Optional properties: + +- VDDD-supply : the regulator provider of VDDD + - micbias-resistor-k-ohms : the bias resistor to be used in kOmhs The resistor can take values of 2k, 4k or 8k. If set to 0 it will be off. @@ -15,17 +23,9 @@ Required properties: - micbias-voltage-m-volts : the bias voltage to be used in mVolts The voltage can take values from 1.25V to 3V by 250mV steps - If this node is not mentionned or the value is unknown, then + If this node is not mentioned or the value is unknown, then the value is set to 1.25V. -- VDDA-supply : the regulator provider of VDDA - -- VDDIO-supply: the regulator provider of VDDIO - -Optional properties: - -- VDDD-supply : the regulator provider of VDDD - Example: codec: sgtl5000@0a { -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: cs53l30: Rename the volume controls for preamplifier
by Nicolin Chen 02 Jun '16

02 Jun '16

Volume controls should end with 'Volume', so this patch renames them for ADC preamplifier. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 714e579..9aff449 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -331,10 +331,10 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_SINGLE_TLV("ADC2 NG Boost Volume", CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_BOOST_SHIFT, 1, 0, adc_ng_boost_tlv), - SOC_DOUBLE_R_TLV("ADC1 Pre Amp Gain", CS53L30_ADC1A_AFE_CTL, + SOC_DOUBLE_R_TLV("ADC1 Preamplifier Volume", CS53L30_ADC1A_AFE_CTL, CS53L30_ADC1B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT, 2, 0, pga_preamp_tlv), - SOC_DOUBLE_R_TLV("ADC2 Pre Amp Gain", CS53L30_ADC2A_AFE_CTL, + SOC_DOUBLE_R_TLV("ADC2 Preamplifier Volume", CS53L30_ADC2A_AFE_CTL, CS53L30_ADC2B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT, 2, 0, pga_preamp_tlv), -- 2.1.4

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[alsa-devel] hda-intel: ALC1150 (alc882 patch) produces constant noise on line-out port
by Maximilian.ammann 02 Jun '16

02 Jun '16

Hello, I experienced weird behaviour on my "ASUS MAXIMUS VII RANGER" which has a realtek soundcard with the ALC1150 codec. The problem is that if the patch_882 is enabled I get a constant noise on the line-out port. I started experienced and noticed that when using the generic model the problem disappeared. Therefore the problem also disappeared when removing the folowing line in sound/pci/hda/patch_realtek.c: HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882), I really would like to know whether this behaviour is expected or how I (myself) could fix it by creating a patch. alsa-info.sh output: Without alc882 patch: http://www.alsa-project.org/db/?f=3fdf926c64a02273e0e68035eeb94282ca44a 7df With alc882 patch: http://www.alsa-project.org/db/?f=336755aab10811bb0af5440dbda7b9cda3d9b 116 Thansk in advance, if possible it would be awesome if you could provide me some tips since I really would like to fix it myself as my very first patch. Max

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Re: [alsa-devel] Applied "ASoC: Intel: add bytct-rt5651 machine driver" to the asoc tree
by Pietro 02 Jun '16

02 Jun '16

Hi, I'm an owner of a 2 in 1 Netbook with an Atom X5 Z8300 SoC with rt5651 codec (Cube iWork 11). To get audio working I Added these lines of code to the ./soc/intel/atom/sst/sst_acpi.c file: --- ./soc/intel/atom/sst/sst_acpi.c.old 2016-05-31 09:05:40.045682194 +0200 +++ ./soc/intel/atom/sst/sst_acpi.c 2016-05-31 09:07:26.829324148 +0200 @@ -342,9 +342,12 @@ &chv_platform_data }, {"193C9890", "cht-bsw-max98090", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, &chv_platform_data }, + {"10EC5651", "bytcr_rt5651", "intel/fw_sst_0f28.bin", "bytcr_rt5651", NULL, + &byt_rvp_platform_data }, {}, }; + static const struct acpi_device_id sst_acpi_ids[] = { { "80860F28", (unsigned long)&sst_acpi_bytcr}, { "808622A8", (unsigned long) &sst_acpi_chv}, Then i added the UCM files linked in these archive: https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 With this configuration, audio seems to work, but is 'slow': Playback speed is at slow motion, and seconds move slowly (Tried with aplay and audacious with many sound files, using directly Alsa and Pulseaudio with no differencies). This seems to be a DSP Clock problem. Have you got any suggestions to solve this problem? Thanks, Pietro

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[alsa-devel] [PATCH] ASoC: tlv320aix31xx: Add ACPI match for Lenovo 100S
by Bastien Nocera 01 Jun '16

01 Jun '16

The Lenovo 100S netbook has a codec controller for which there is a driver, but doesn't know how to access the device. This adds the necessary ACPI table for the driver to find the device. Device (TTLV) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10TI3100") // _HID: Hardware ID Name (_CID, "10TI3100") // _CID: Compatible ID Name (_DDN, "TI TLV320AIC3100 Codec Controller ") // _DDN: DOS Device Name Name (_UID, One) // _UID: Unique ID Signed-off-by: Bastien Nocera <hadess(a)hadess.net> Tested-by: Jan Schmidt <jan(a)centricular.com> --- sound/soc/codecs/tlv320aic31xx.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index ee4def4..3c5e1df 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -28,6 +28,7 @@ #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/acpi.h> #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/slab.h> @@ -1280,10 +1281,19 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +#ifdef CONFIG_ACPI +static const struct acpi_device_id aic31xx_acpi_match[] = { + { "10TI3100", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); +#endif + static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + .acpi_match_table = ACPI_PTR(aic31xx_acpi_match), }, .probe = aic31xx_i2c_probe, .remove = aic31xx_i2c_remove, -- 2.7.3

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[alsa-devel] [PATCH v2] ASoC: tas571x: add input channel mixer for TAS5717/19
by Petr Kulhavy 01 Jun '16

01 Jun '16

Add channel 1 and 2 input mixer registers and the related ALSA mixer controls for TAS5717/19 chips. The mixer control coefficients on the chip are linear in the range -3.99999 to +3.99999, encoded in 3.23 number format. In this patch the mixer controls are limited to 128 values from 0.0 to 1.0 in 1/64 steps. Signed-off-by: Petr Kulhavy <brain(a)jikos.cz> --- v1: initial v2: re-apply on topics/tas571x sound/soc/codecs/tas571x.c | 18 ++++++++++++++++++ sound/soc/codecs/tas571x.h | 5 +++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index bc1fbaf..d8baca3 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -64,6 +64,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_INPUT_MUX_REG: case TAS571X_CH4_SRC_SELECT_REG: case TAS571X_PWM_MUX_REG: + case TAS5717_CH1_RIGHT_CH_MIX_REG: + case TAS5717_CH1_LEFT_CH_MIX_REG: + case TAS5717_CH2_LEFT_CH_MIX_REG: + case TAS5717_CH2_RIGHT_CH_MIX_REG: return 4; default: return 1; @@ -397,6 +401,16 @@ static const struct snd_kcontrol_new tas5711_controls[] = { TAS571X_SOFT_MUTE_REG, TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, 1, 1), + + SOC_DOUBLE_R_RANGE("CH1 Mixer Volume", + TAS5717_CH1_LEFT_CH_MIX_REG, + TAS5717_CH1_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), + + SOC_DOUBLE_R_RANGE("CH2 Mixer Volume", + TAS5717_CH2_LEFT_CH_MIX_REG, + TAS5717_CH2_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), }; static const struct regmap_range tas571x_readonly_regs_range[] = { @@ -520,6 +534,10 @@ static const struct reg_default tas5717_reg_defaults[] = { { 0x08, 0x00c0 }, { 0x09, 0x00c0 }, { 0x1b, 0x82 }, + { TAS5717_CH1_RIGHT_CH_MIX_REG, 0x0 }, + { TAS5717_CH1_LEFT_CH_MIX_REG, 0x800000}, + { TAS5717_CH2_LEFT_CH_MIX_REG, 0x0 }, + { TAS5717_CH2_RIGHT_CH_MIX_REG, 0x800000}, }; static const struct regmap_config tas5717_regmap_config = { diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index bf4d436..c45677b 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -87,4 +87,9 @@ #define TAS5717_CH3_BQ0_REG 0x5e #define TAS5717_CH3_BQ1_REG 0x5f +#define TAS5717_CH1_RIGHT_CH_MIX_REG 0x72 +#define TAS5717_CH1_LEFT_CH_MIX_REG 0x73 +#define TAS5717_CH2_LEFT_CH_MIX_REG 0x76 +#define TAS5717_CH2_RIGHT_CH_MIX_REG 0x77 + #endif /* _TAS571X_H */ -- 2.5.0

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