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[alsa-devel] [GIT PULL] MOP500 Audio patches due for ASoC
by Lee Jones 20 Sep '12

20 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.linaro.org/people/ljones/linux-3.0-ux500.git for-asoc-next for you to fetch changes up to b4cad7af6665743647f28119b689e1552326d4da: Documentation: Define the MSP Driver Device Tree bindings (2012-09-20 09:10:59 +0200) ---------------------------------------------------------------- Lee Jones (7): ASoC: Ux500: Move MSP pinctrl setup into the MSP driver ASoC: Ux500: Enable MOP500 driver for Device Tree ASoC: Ux500: Enable ux500 MSP driver for Device Tree ASoC: codecs: Enable AB8500 CODEC for Device Tree ASoC: Ux500: Minor coding layout changes Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings Documentation: Define the MSP Driver Device Tree bindings .../devicetree/bindings/sound/ux500-mop500.txt | 39 +++++++++ .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++++++++ arch/arm/mach-ux500/board-mop500-msp.c | 79 +---------------- arch/arm/mach-ux500/include/mach/msp.h | 2 - include/linux/mfd/abx500/ab8500-codec.h | 6 +- sound/soc/codecs/ab8500-codec.c | 81 ++++++++++++++++++ sound/soc/ux500/mop500.c | 47 +++++++++-- sound/soc/ux500/ux500_msp_dai.c | 6 ++ sound/soc/ux500/ux500_msp_i2s.c | 89 +++++++++++++++----- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 10 files changed, 292 insertions(+), 108 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt -- Lee Jones Linaro ST-Ericsson Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

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[alsa-devel] [PATCH v2] ASoC: codecs: Add DA9055 codec driver
by Ashish Chavan 20 Sep '12

20 Sep '12

This patch adds support for Dialog semiconductor's DA9055 audio codec. This has been tested on DA9055 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- Changes since v1: - Made extensive use of DAPM throughout the driver - Removed LDO, DMIC and other non used controls - Corrected names for many controls - Other minor improvements --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9055.c | 1468 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1474 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3684255..b92759a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DA732X if I2C + select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC @@ -239,6 +240,9 @@ config SND_SOC_DA7210 config SND_SOC_DA732X tristate +config SND_SOC_DA9055 + tristate + config SND_SOC_DFBMCS320 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ca508b2..9bd4d95 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da732x-objs := da732x.o +snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o @@ -144,6 +145,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o +obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c new file mode 100644 index 0000000..86661ec --- /dev/null +++ b/sound/soc/codecs/da9055.c @@ -0,0 +1,1468 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +/* DA9055 register space */ + +/* Status Registers */ +#define DA9055_STATUS1 0x02 +#define DA9055_PLL_STATUS 0x03 +#define DA9055_AUX_L_GAIN_STATUS 0x04 +#define DA9055_AUX_R_GAIN_STATUS 0x05 +#define DA9055_MIC_L_GAIN_STATUS 0x06 +#define DA9055_MIC_R_GAIN_STATUS 0x07 +#define DA9055_MIXIN_L_GAIN_STATUS 0x08 +#define DA9055_MIXIN_R_GAIN_STATUS 0x09 +#define DA9055_ADC_L_GAIN_STATUS 0x0A +#define DA9055_ADC_R_GAIN_STATUS 0x0B +#define DA9055_DAC_L_GAIN_STATUS 0x0C +#define DA9055_DAC_R_GAIN_STATUS 0x0D +#define DA9055_HP_L_GAIN_STATUS 0x0E +#define DA9055_HP_R_GAIN_STATUS 0x0F +#define DA9055_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA9055_CIF_CTRL 0x20 +#define DA9055_DIG_ROUTING_AIF 0X21 +#define DA9055_SR 0x22 +#define DA9055_REFERENCES 0x23 +#define DA9055_PLL_FRAC_TOP 0x24 +#define DA9055_PLL_FRAC_BOT 0x25 +#define DA9055_PLL_INTEGER 0x26 +#define DA9055_PLL_CTRL 0x27 +#define DA9055_AIF_CLK_MODE 0x28 +#define DA9055_AIF_CTRL 0x29 +#define DA9055_DIG_ROUTING_DAC 0x2A +#define DA9055_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA9055_AUX_L_GAIN 0x30 +#define DA9055_AUX_R_GAIN 0x31 +#define DA9055_MIXIN_L_SELECT 0x32 +#define DA9055_MIXIN_R_SELECT 0x33 +#define DA9055_MIXIN_L_GAIN 0x34 +#define DA9055_MIXIN_R_GAIN 0x35 +#define DA9055_ADC_L_GAIN 0x36 +#define DA9055_ADC_R_GAIN 0x37 +#define DA9055_ADC_FILTERS1 0x38 +#define DA9055_MIC_L_GAIN 0x39 +#define DA9055_MIC_R_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA9055_DAC_FILTERS5 0x40 +#define DA9055_DAC_FILTERS2 0x41 +#define DA9055_DAC_FILTERS3 0x42 +#define DA9055_DAC_FILTERS4 0x43 +#define DA9055_DAC_FILTERS1 0x44 +#define DA9055_DAC_L_GAIN 0x45 +#define DA9055_DAC_R_GAIN 0x46 +#define DA9055_CP_CTRL 0x47 +#define DA9055_HP_L_GAIN 0x48 +#define DA9055_HP_R_GAIN 0x49 +#define DA9055_LINE_GAIN 0x4A +#define DA9055_MIXOUT_L_SELECT 0x4B +#define DA9055_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA9055_SYSTEM_MODES_INPUT 0x50 +#define DA9055_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA9055_AUX_L_CTRL 0x60 +#define DA9055_AUX_R_CTRL 0x61 +#define DA9055_MIC_BIAS_CTRL 0x62 +#define DA9055_MIC_L_CTRL 0x63 +#define DA9055_MIC_R_CTRL 0x64 +#define DA9055_MIXIN_L_CTRL 0x65 +#define DA9055_MIXIN_R_CTRL 0x66 +#define DA9055_ADC_L_CTRL 0x67 +#define DA9055_ADC_R_CTRL 0x68 +#define DA9055_DAC_L_CTRL 0x69 +#define DA9055_DAC_R_CTRL 0x6A +#define DA9055_HP_L_CTRL 0x6B +#define DA9055_HP_R_CTRL 0x6C +#define DA9055_LINE_CTRL 0x6D +#define DA9055_MIXOUT_L_CTRL 0x6E +#define DA9055_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA9055_LDO_CTRL 0x90 +#define DA9055_IO_CTRL 0x91 +#define DA9055_GAIN_RAMP_CTRL 0x92 +#define DA9055_MIC_CONFIG 0x93 +#define DA9055_PC_COUNT 0x94 +#define DA9055_CP_VOL_THRESHOLD1 0x95 +#define DA9055_CP_DELAY 0x96 +#define DA9055_CP_DETECTOR 0x97 +#define DA9055_AIF_OFFSET 0x98 +#define DA9055_DIG_CTRL 0x99 +#define DA9055_ALC_CTRL2 0x9A +#define DA9055_ALC_CTRL3 0x9B +#define DA9055_ALC_NOISE 0x9C +#define DA9055_ALC_TARGET_MIN 0x9D +#define DA9055_ALC_TARGET_MAX 0x9E +#define DA9055_ALC_GAIN_LIMITS 0x9F +#define DA9055_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA9055_ALC_ANTICLIP_CTRL 0xA1 +#define DA9055_ALC_ANTICLIP_LEVEL 0xA2 +#define DA9055_ALC_OFFSET_OP2M_L 0xA6 +#define DA9055_ALC_OFFSET_OP2U_L 0xA7 +#define DA9055_ALC_OFFSET_OP2M_R 0xAB +#define DA9055_ALC_OFFSET_OP2U_R 0xAC +#define DA9055_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA9055_ALC_CIC_OP_LVL_DATA 0xAE +#define DA9055_DAC_NG_SETUP_TIME 0xAF +#define DA9055_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA9055_DAC_NG_ON_THRESHOLD 0xB1 +#define DA9055_DAC_NG_CTRL 0xB2 + +/* SR bit fields */ +#define DA9055_SR_8000 (0x1 << 0) +#define DA9055_SR_11025 (0x2 << 0) +#define DA9055_SR_12000 (0x3 << 0) +#define DA9055_SR_16000 (0x5 << 0) +#define DA9055_SR_22050 (0x6 << 0) +#define DA9055_SR_24000 (0x7 << 0) +#define DA9055_SR_32000 (0x9 << 0) +#define DA9055_SR_44100 (0xA << 0) +#define DA9055_SR_48000 (0xB << 0) +#define DA9055_SR_88200 (0xE << 0) +#define DA9055_SR_96000 (0xF << 0) + +/* REFERENCES bit fields */ +#define DA9055_BIAS_EN (1 << 3) +#define DA9055_VMID_EN (1 << 7) + +/* PLL_CTRL bit fields */ +#define DA9055_PLL_INDIV_10_20_MHZ (1 << 2) +#define DA9055_PLL_SRM_EN (1 << 6) +#define DA9055_PLL_EN (1 << 7) + +/* AIF_CLK_MODE bit fields */ +#define DA9055_AIF_BCLKS_PER_WCLK_32 (0 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_64 (1 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_128 (2 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_256 (3 << 0) +#define DA9055_AIF_CLK_EN_SLAVE_MODE (0 << 7) +#define DA9055_AIF_CLK_EN_MASTER_MODE (1 << 7) + +/* AIF_CTRL bit fields */ +#define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) +#define DA9055_AIF_FORMAT_LEFT_J (1 << 0) +#define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_WORD_S16_LE (0 << 2) +#define DA9055_AIF_WORD_S20_3LE (1 << 2) +#define DA9055_AIF_WORD_S24_LE (2 << 2) +#define DA9055_AIF_WORD_S32_LE (3 << 2) + +/* MIXIN_L_CTRL bit fields */ +#define DA9055_MIXIN_L_MIX_EN (1 << 3) + +/* MIXIN_R_CTRL bit fields */ +#define DA9055_MIXIN_R_MIX_EN (1 << 3) + +/* ADC_L_CTRL bit fields */ +#define DA9055_ADC_L_EN (1 << 7) + +/* ADC_R_CTRL bit fields */ +#define DA9055_ADC_R_EN (1 << 7) + +/* DAC_L_CTRL bit fields */ +#define DA9055_DAC_L_MUTE_EN (1 << 6) + +/* DAC_R_CTRL bit fields */ +#define DA9055_DAC_R_MUTE_EN (1 << 6) + +/* HP_L_CTRL bit fields */ +#define DA9055_HP_L_AMP_OE (1 << 3) + +/* HP_R_CTRL bit fields */ +#define DA9055_HP_R_AMP_OE (1 << 3) + +/* LINE_CTRL bit fields */ +#define DA9055_LINE_AMP_OE (1 << 3) + +/* MIXOUT_L_CTRL bit fields */ +#define DA9055_MIXOUT_L_MIX_EN (1 << 3) + +/* MIXOUT_R_CTRL bit fields */ +#define DA9055_MIXOUT_R_MIX_EN (1 << 3) + +/* ALC_CIC_OP_LEVEL_CTRL bit fields */ +#define DA9055_ALC_DATA_MIDDLE (2 << 0) +#define DA9055_ALC_DATA_TOP (3 << 0) +#define DA9055_ALC_CIC_OP_CHANNEL_LEFT (0 << 7) +#define DA9055_ALC_CIC_OP_CHANNEL_RIGHT (1 << 7) + +#define DA9055_AIF_BCLK_MASK (3 << 0) +#define DA9055_AIF_CLK_MODE_MASK (1 << 7) +#define DA9055_AIF_FORMAT_MASK (3 << 0) +#define DA9055_AIF_WORD_LENGTH_MASK (3 << 2) +#define DA9055_GAIN_RAMPING_EN (1 << 5) + +#define DA9055_ALC_OFFSET_15_8 0x00FF00 +#define DA9055_ALC_OFFSET_17_16 0x030000 +#define DA9055_ALC_AVG_ITERATIONS 5 + +struct pll_div { + int fref; + int fout; + u8 frac_top; + u8 frac_bot; + u8 integer; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL divisor table */ +static const struct pll_div da9055_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz and its harmonics */ + {11289600, 2822400, 0x00, 0x00, 0x20, 1}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x03, 0x61, 0x1E, 1}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x0C, 0xCC, 0x1D, 1}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x19, 0x45, 0x1B, 1}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x18, 0x56, 0x1A, 1}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x02, 0xD0, 0x19, 1}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x1A, 0x1C, 0x12, 1}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x0B, 0x6D, 0x12, 1}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x07, 0xDD, 0x12, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz and its harmonics */ + {11289600, 3072000, 0x1A, 0x8E, 0x22, 1}, /* MCLK=11.2896Mhz */ + {12000000, 3072000, 0x18, 0x93, 0x20, 1}, /* MCLK=12Mhz */ + {12288000, 3072000, 0x00, 0x00, 0x20, 1}, /* MCLK=12.288Mhz */ + {13000000, 3072000, 0x07, 0xEA, 0x1E, 1}, /* MCLK=13Mhz */ + {13500000, 3072000, 0x04, 0x11, 0x1D, 1}, /* MCLK=13.5Mhz */ + {14400000, 3072000, 0x09, 0xD0, 0x1B, 1}, /* MCLK=14.4Mhz */ + {19200000, 3072000, 0x0F, 0x5C, 0x14, 1}, /* MCLK=19.2Mhz */ + {19680000, 3072000, 0x1F, 0x60, 0x13, 1}, /* MCLK=19.68Mhz */ + {19800000, 3072000, 0x1B, 0x80, 0x13, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + {11289600, 2822400, 0x0D, 0x47, 0x21, 0}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x0D, 0xFA, 0x1F, 0}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x16, 0x66, 0x1E, 0}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x00, 0x98, 0x1D, 0}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x1E, 0x33, 0x1B, 0}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x06, 0x50, 0x1A, 0}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x14, 0xBC, 0x13, 0}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x05, 0x66, 0x13, 0}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x01, 0xAE, 0x13, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA9055_CLKSRC_MCLK +}; + +/* Gain and Volume */ + +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da9055_hpf_cutoff_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da9055_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +static const struct soc_enum da9055_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da9055_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da9055_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +static const struct soc_enum da9055_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +/* Gain ramping rate value */ +static const char * const da9055_gain_ramping_txt[] = { + "nominal rate", "nominal rate * 4", "nominal rate * 8", + "nominal rate / 8" +}; + +static const struct soc_enum da9055_gain_ramping_rate = + SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); + +/* DAC noise gate setup time value */ +static const char * const da9055_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da9055_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, + da9055_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da9055_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, + da9055_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da9055_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, + da9055_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da9055_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da9055_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, + da9055_dac_soft_mute_rate_txt); + +/* DAC routing select */ +static const char * const da9055_dac_select_txt[] = { + "ADC left output", "ADC right output", "AIF input left", + "AIF input right" +}; + +static const struct soc_enum da9055_dac_l_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_select_txt); + +static const struct soc_enum da9055_dac_r_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_select_txt); + +/* MIC bias select */ +static const char * const da9055_mic_bias_select_txt[] = { + "MIC Bias 1", "MIC Bias 2" +}; + +static const struct soc_enum da9055_mic_bias_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 6, 2, + da9055_mic_bias_select_txt); + +/* MIC PGA Left source select */ +static const char * const da9055_mic_l_src_txt[] = { + "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_l_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt); + +/* MIC PGA Right source select */ +static const char * const da9055_mic_r_src_txt[] = { + "MIC2_R_L", "MIC2_R", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_r_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da9055_signal_tracking_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da9055_integ_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, + da9055_signal_tracking_rate_txt); + +static const struct soc_enum da9055_integ_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, + da9055_signal_tracking_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da9055_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da9055_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da9055_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da9055_hold_time = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); + +static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA9055_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA9055_ALC_AVG_ITERATIONS; +} + +static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 reg_val, adc_left, adc_right; + int avg_left_data, avg_right_data, offset_l, offset_r; + + if (ucontrol->value.integer.value[0]) { + /* + * While enabling ALC (or ALC sync mode), calibration of the DC + * offsets must be done first + */ + + /* Save current values from ADC control registers */ + adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); + adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_ADC_L_EN, DA9055_ADC_L_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_ADC_R_EN, DA9055_ADC_R_EN); + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_L, reg_val); + reg_val = (offset_l & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_L, reg_val); + + reg_val = (offset_r & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_R, reg_val); + reg_val = (offset_r & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_R, reg_val); + + /* Restore original values of ADC control registers */ + snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); + snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +static const struct snd_kcontrol_new da9055_snd_controls[] = { + + /* Volume controls */ + SOC_DOUBLE_R_TLV("Mic Volume", + DA9055_MIC_L_GAIN, DA9055_MIC_R_GAIN, + 0, 0x7, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", + DA9055_AUX_L_GAIN, DA9055_AUX_R_GAIN, + 0, 0x3f, 0, aux_vol_tlv), + SOC_DOUBLE_R_TLV("Mixin PGA Volume", + DA9055_MIXIN_L_GAIN, DA9055_MIXIN_R_GAIN, + 0, 0xf, 0, mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", + DA9055_ADC_L_GAIN, DA9055_ADC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Volume", + DA9055_DAC_L_GAIN, DA9055_DAC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", + DA9055_HP_L_GAIN, DA9055_HP_R_GAIN, + 0, 0x3f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA9055_LINE_GAIN, 0, 0x3f, 0, + lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA9055_DAC_FILTERS4, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA9055_DAC_FILTERS2, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA9055_DAC_FILTERS2, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA9055_DAC_FILTERS3, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA9055_DAC_FILTERS3, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA9055_DAC_FILTERS4, 0, 0xf, 0, + eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA9055_ADC_FILTERS1, 7, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da9055_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA9055_ADC_FILTERS1, 3, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da9055_adc_vf_cutoff), + + SOC_SINGLE("DAC HPF Switch", DA9055_DAC_FILTERS1, 7, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da9055_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA9055_DAC_FILTERS1, 3, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da9055_dac_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Switch", DA9055_MIC_L_CTRL, + DA9055_MIC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Aux Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Mixin PGA Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("ADC Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Headphone Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 6, 1, 0), + SOC_SINGLE("Lineout Switch", DA9055_LINE_CTRL, 6, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA9055_DAC_FILTERS5, 7, 1, 0), + SOC_ENUM("DAC Soft Mute Rate", da9055_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Headphone ZC Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 4, 1, 0), + SOC_SINGLE("Lineout ZC Switch", DA9055_LINE_CTRL, 4, 1, 0), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA9055_DAC_L_CTRL, + DA9055_DAC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Headphone Gain Ramping Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 5, 1, 0), + SOC_SINGLE("Lineout Gain Ramping Switch", DA9055_LINE_CTRL, 5, 1, 0), + SOC_ENUM("Gain Ramping Rate", da9055_gain_ramping_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Switch", DA9055_DAC_NG_CTRL, 7, 1, 0), + SOC_SINGLE("DAC NG ON Threshold", DA9055_DAC_NG_ON_THRESHOLD, + 0, 0x7, 0), + SOC_SINGLE("DAC NG OFF Threshold", DA9055_DAC_NG_OFF_THRESHOLD, + 0, 0x7, 0), + SOC_ENUM("DAC NG Setup Time", da9055_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da9055_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da9055_dac_ng_rampdown_rate), + + /* DAC Routing control */ + SOC_ENUM("DAC Left Select", da9055_dac_l_select), + SOC_ENUM("DAC Right Select", da9055_dac_r_select), + + /* DAC Invertion control */ + SOC_SINGLE("DAC Left Invert", DA9055_DIG_CTRL, 3, 1, 0), + SOC_SINGLE("DAC Right Invert", DA9055_DIG_CTRL, 7, 1, 0), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Switch", DA9055_MIXIN_L_SELECT, + DA9055_MIXIN_R_SELECT, 7, 1, 0), + + /* To select if MIC Bias powers MIC 1 or MIC 2 */ + SOC_ENUM("Mic Bias Select", da9055_mic_bias_select), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Switch", DA9055_ALC_CTRL1, 3, 7, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE_EXT("ALC Sync Mode Switch", DA9055_ALC_CTRL1, 1, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE("ALC Offset Switch", DA9055_ALC_CTRL1, 0, 1, 0), + SOC_SINGLE("ALC Anticlip Mode Switch", DA9055_ALC_ANTICLIP_CTRL, + 7, 1, 0), + SOC_SINGLE("ALC Anticlip Level", DA9055_ALC_ANTICLIP_LEVEL, + 0, 0x7f, 0), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA9055_ALC_TARGET_MIN, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA9055_ALC_TARGET_MAX, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA9055_ALC_NOISE, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA9055_ALC_GAIN_LIMITS, + 4, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA9055_ALC_GAIN_LIMITS, + 0, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 0, 0x7, 0, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 4, 0x7, 0, alc_analog_gain_tlv), + SOC_ENUM("ALC Attack Rate", da9055_attack_rate), + SOC_ENUM("ALC Release Rate", da9055_release_rate), + SOC_ENUM("ALC Hold Time", da9055_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da9055_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da9055_integ_release_rate), +}; + +/* DAPM Controls */ + +/* Mic PGA Left Source */ +static const struct snd_kcontrol_new da9055_mic_l_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_l_src); + +/* Mic PGA Right Source */ +static const struct snd_kcontrol_new da9055_mic_r_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_r_src); + +/* In Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXIN_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_L_SELECT, 2, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXIN_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXIN_R_SELECT, 3, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXOUT_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_L_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA9055_MIXOUT_L_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_L_SELECT, + 6, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXOUT_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA9055_MIXOUT_R_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_R_SELECT, + 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { + /* Input Side */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* MUXs for Mic PGA source selection */ + SND_SOC_DAPM_MUX("Mic Left Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_l_mux_controls), + SND_SOC_DAPM_MUX("Mic Right Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_r_mux_controls), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA9055_MIC_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA9055_MIC_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left", DA9055_AUX_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right", DA9055_AUX_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Left", DA9055_MIXIN_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Right", DA9055_MIXIN_R_CTRL, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", DA9055_MIC_BIAS_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF", DA9055_AIF_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", DA9055_CP_CTRL, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinl_controls[0], + ARRAY_SIZE(da9055_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinr_controls[0], + ARRAY_SIZE(da9055_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA9055_ADC_L_CTRL, 7, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA9055_ADC_R_CTRL, 7, 0), + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA9055_DAC_L_CTRL, 7, 0), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA9055_DAC_R_CTRL, 7, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutl_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutr_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("MIXOUT Left", DA9055_MIXOUT_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXOUT Right", DA9055_MIXOUT_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout", DA9055_LINE_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA9055_HP_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA9055_HP_R_CTRL, 7, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da9055_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"Mic Left Source", "MIC1_P_N", "MIC1"}, + {"Mic Left Source", "MIC1_P", "MIC1"}, + {"Mic Left Source", "MIC1_N", "MIC1"}, + {"Mic Left Source", "MIC2_L", "MIC2"}, + + {"Mic Right Source", "MIC2_R_L", "MIC2"}, + {"Mic Right Source", "MIC2_R", "MIC2"}, + {"Mic Right Source", "MIC2_L", "MIC2"}, + + {"Mic Left", NULL, "Mic Left Source"}, + {"Mic Right", NULL, "Mic Right Source"}, + + {"Aux Left", NULL, "AUXL"}, + {"Aux Right", NULL, "AUXR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux Left Switch", "Aux Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux Right Switch", "Aux Right"}, + {"In Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + + {"MIXIN Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "MIXIN Left"}, + + {"MIXIN Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "MIXIN Right"}, + + {"ADC Left", NULL, "AIF"}, + {"ADC Right", NULL, "AIF"}, + + /* Output path */ + {"DAC Left", NULL, "AIF"}, + {"DAC Right", NULL, "AIF"}, + + {"Out Mixer Left", "Aux Left Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Left", "Aux Left Invert Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux Right Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Right", "Aux Right Invert Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"MIXOUT Left", NULL, "Out Mixer Left"}, + {"Headphone Left", NULL, "MIXOUT Left"}, + {"Headphone Left", NULL, "Charge Pump"}, + {"HPL", NULL, "Headphone Left"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Headphone Right", NULL, "MIXOUT Right"}, + {"Headphone Right", NULL, "Charge Pump"}, + {"HPR", NULL, "Headphone Right"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Lineout", NULL, "MIXOUT Right"}, + {"LINE", NULL, "Lineout"}, +}; + +/* Codec private data */ +struct da9055_priv { + struct regmap *regmap; + unsigned int mclk_rate; + int master; +}; + +static struct reg_default da9055_reg_defaults[] = { + { 0x21, 0x10 }, + { 0x22, 0x0A }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x0C }, + { 0x28, 0x01 }, + { 0x29, 0x08 }, + { 0x2A, 0x32 }, + { 0x2B, 0x00 }, + { 0x30, 0x35 }, + { 0x31, 0x35 }, + { 0x32, 0x00 }, + { 0x33, 0x00 }, + { 0x34, 0x03 }, + { 0x35, 0x03 }, + { 0x36, 0x6F }, + { 0x37, 0x6F }, + { 0x38, 0x80 }, + { 0x39, 0x01 }, + { 0x3A, 0x01 }, + { 0x40, 0x00 }, + { 0x41, 0x88 }, + { 0x42, 0x88 }, + { 0x43, 0x08 }, + { 0x44, 0x80 }, + { 0x45, 0x6F }, + { 0x46, 0x6F }, + { 0x47, 0x61 }, + { 0x48, 0x35 }, + { 0x49, 0x35 }, + { 0x4A, 0x35 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x60, 0x44 }, + { 0x61, 0x44 }, + { 0x62, 0x00 }, + { 0x63, 0x40 }, + { 0x64, 0x40 }, + { 0x65, 0x40 }, + { 0x66, 0x40 }, + { 0x67, 0x40 }, + { 0x68, 0x40 }, + { 0x69, 0x48 }, + { 0x6A, 0x40 }, + { 0x6B, 0x41 }, + { 0x6C, 0x40 }, + { 0x6D, 0x40 }, + { 0x6E, 0x10 }, + { 0x6F, 0x10 }, + { 0x90, 0x80 }, + { 0x92, 0x02 }, + { 0x93, 0x00 }, + { 0x99, 0x00 }, + { 0x9A, 0x00 }, + { 0x9B, 0x00 }, + { 0x9C, 0x3F }, + { 0x9D, 0x00 }, + { 0x9E, 0x3F }, + { 0x9F, 0xFF }, + { 0xA0, 0x71 }, + { 0xA1, 0x00 }, + { 0xA2, 0x00 }, + { 0xA6, 0x00 }, + { 0xA7, 0x00 }, + { 0xAB, 0x00 }, + { 0xAC, 0x00 }, + { 0xAD, 0x00 }, + { 0xAF, 0x08 }, + { 0xB0, 0x00 }, + { 0xB1, 0x00 }, + { 0xB2, 0x00 }, +}; + +static bool da9055_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case DA9055_STATUS1: + case DA9055_PLL_STATUS: + case DA9055_AUX_L_GAIN_STATUS: + case DA9055_AUX_R_GAIN_STATUS: + case DA9055_MIC_L_GAIN_STATUS: + case DA9055_MIC_R_GAIN_STATUS: + case DA9055_MIXIN_L_GAIN_STATUS: + case DA9055_MIXIN_R_GAIN_STATUS: + case DA9055_ADC_L_GAIN_STATUS: + case DA9055_ADC_R_GAIN_STATUS: + case DA9055_DAC_L_GAIN_STATUS: + case DA9055_DAC_R_GAIN_STATUS: + case DA9055_HP_L_GAIN_STATUS: + case DA9055_HP_R_GAIN_STATUS: + case DA9055_LINE_GAIN_STATUS: + case DA9055_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +/* Set DAI word length */ +static int da9055_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_ctrl, fs; + u32 sysclk; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif_ctrl = DA9055_AIF_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif_ctrl = DA9055_AIF_WORD_S20_3LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif_ctrl = DA9055_AIF_WORD_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif_ctrl = DA9055_AIF_WORD_S32_LE; + break; + default: + return -EINVAL; + } + + /* Set AIF format */ + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_WORD_LENGTH_MASK, + aif_ctrl); + + switch (params_rate(params)) { + case 8000: + fs = DA9055_SR_8000; + sysclk = 3072000; + break; + case 11025: + fs = DA9055_SR_11025; + sysclk = 2822400; + break; + case 12000: + fs = DA9055_SR_12000; + sysclk = 3072000; + break; + case 16000: + fs = DA9055_SR_16000; + sysclk = 3072000; + break; + case 22050: + fs = DA9055_SR_22050; + sysclk = 2822400; + break; + case 32000: + fs = DA9055_SR_32000; + sysclk = 3072000; + break; + case 44100: + fs = DA9055_SR_44100; + sysclk = 2822400; + break; + case 48000: + fs = DA9055_SR_48000; + sysclk = 3072000; + break; + case 88200: + fs = DA9055_SR_88200; + sysclk = 2822400; + break; + case 96000: + fs = DA9055_SR_96000; + sysclk = 3072000; + break; + default: + return -EINVAL; + } + + if (da9055->mclk_rate) { + /* PLL Mode, Write actual FS */ + snd_soc_write(codec, DA9055_SR, fs); + } else { + /* + * Non-PLL Mode + * When PLL is bypassed, chip assumes constant MCLK of + * 12.288MHz and uses sample rate value to divide this MCLK + * to derive its sys clk. As sys clk has to be 256 * Fs, we + * need to write constant sample rate i.e. 48KHz. + */ + snd_soc_write(codec, DA9055_SR, DA9055_SR_48000); + } + + if (da9055->mclk_rate && (da9055->mclk_rate != sysclk)) { + /* PLL Mode */ + if (!da9055->master) { + /* PLL slave mode, enable PLL and also SRM */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN | DA9055_PLL_SRM_EN, + DA9055_PLL_EN | DA9055_PLL_SRM_EN); + } else { + /* PLL master mode, only enable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN, DA9055_PLL_EN); + } + } else { + /* Non PLL Mode, disable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + } + + return 0; +} + +/* Set DAI mode and Format */ +static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_clk_mode, aif_ctrl, mode; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* DA9055 in I2S Master Mode */ + mode = 1; + aif_clk_mode = DA9055_AIF_CLK_EN_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* DA9055 in I2S Slave Mode */ + mode = 0; + aif_clk_mode = DA9055_AIF_CLK_EN_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* Don't allow change of mode if PLL is enabled */ + if ((snd_soc_read(codec, DA9055_PLL_CTRL) & DA9055_PLL_EN) && + (da9055->master != mode)) + return -EINVAL; + + da9055->master = mode; + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_ctrl = DA9055_AIF_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_ctrl = DA9055_AIF_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + aif_clk_mode |= DA9055_AIF_BCLKS_PER_WCLK_32; + + snd_soc_update_bits(codec, DA9055_AIF_CLK_MODE, + (DA9055_AIF_CLK_MODE_MASK | DA9055_AIF_BCLK_MASK), + aif_clk_mode); + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_FORMAT_MASK, + aif_ctrl); + return 0; +} + +static int da9055_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, DA9055_DAC_R_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, 0); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, 0); + } + + return 0; +} + +#define DA9055_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da9055_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA9055_CLKSRC_MCLK: + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da9055->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* + * da9055_set_dai_pll : Configure the codec PLL + * @param codec_dai : Pointer to codec DAI + * @param pll_id : da9055 has only one pll, so pll_id is always zero + * @param fref : Input MCLK frequency + * @param fout : FsDM value + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 11.2896MHz, 12MHz, 12.288MHz, + * 13MHz, 13.5MHz, 14.4MHz, 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da9055_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + u8 pll_frac_top, pll_frac_bot, pll_integer, cnt; + + /* Disable PLL before setting the divisors */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + + /* In slave mode, there is only one set of divisors */ + if (!da9055->master && (fout != 2822400)) + goto pll_err; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da9055_pll_div); cnt++) { + /* Check fref, mode and fout */ + if ((fref == da9055_pll_div[cnt].fref) && + (da9055->master == da9055_pll_div[cnt].mode) && + (fout == da9055_pll_div[cnt].fout)) { + /* All match, pick up divisors */ + pll_frac_top = da9055_pll_div[cnt].frac_top; + pll_frac_bot = da9055_pll_div[cnt].frac_bot; + pll_integer = da9055_pll_div[cnt].integer; + break; + } + } + if (cnt >= ARRAY_SIZE(da9055_pll_div)) + goto pll_err; + + /* Write PLL dividers */ + snd_soc_write(codec, DA9055_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA9055_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA9055_PLL_INTEGER, pll_integer); + + return 0; +pll_err: + dev_err(codec_dai->dev, "Error in setting up PLL\n"); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da9055_dai_ops = { + .hw_params = da9055_hw_params, + .set_fmt = da9055_set_dai_fmt, + .set_sysclk = da9055_set_dai_sysclk, + .set_pll = da9055_set_dai_pll, + .digital_mute = da9055_mute, +}; + +static struct snd_soc_dai_driver da9055_dai = { + .name = "da9055-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + .ops = &da9055_dai_ops, + .symmetric_rates = 1, +}; + +static int da9055_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Enable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, + DA9055_VMID_EN | DA9055_BIAS_EN); + } + regcache_sync(da9055->regmap); + break; + case SND_SOC_BIAS_OFF: + /* Disable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int da9055_probe(struct snd_soc_codec *codec) +{ + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = da9055->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + da9055->mclk_rate = 0; /* This will be set from set_sysclk() */ + da9055->master = 0; /* This will be set from set_fmt() */ + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_AUX_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + + /* + * There are two separate control bits for input and output mixers as + * well as headphone and line outs. + * One to enable corresponding amplifier and other to enable its + * output. As amplifier bits are related to power control, they are + * being managed by DAPM while other (non power related) bits are + * enabled here + */ + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_L_MIX_EN, DA9055_MIXIN_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_MIXIN_R_MIX_EN, DA9055_MIXIN_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_MIX_EN, DA9055_MIXOUT_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_MIX_EN, DA9055_MIXOUT_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_HP_L_AMP_OE, DA9055_HP_L_AMP_OE); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_HP_R_AMP_OE, DA9055_HP_R_AMP_OE); + + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_LINE_AMP_OE, DA9055_LINE_AMP_OE); + + /* Set this as per your system configuration */ + snd_soc_write(codec, DA9055_PLL_CTRL, DA9055_PLL_INDIV_10_20_MHZ); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da9055 = { + .probe = da9055_probe, + .set_bias_level = da9055_set_bias_level, + + .controls = da9055_snd_controls, + .num_controls = ARRAY_SIZE(da9055_snd_controls), + + .dapm_widgets = da9055_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da9055_dapm_widgets), + .dapm_routes = da9055_audio_map, + .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), +}; + +static const struct regmap_config da9055_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da9055_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da9055_reg_defaults), + .volatile_reg = da9055_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da9055_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da9055_priv *da9055; + int ret; + + da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + GFP_KERNEL); + if (!da9055) + return -ENOMEM; + + i2c_set_clientdata(i2c, da9055); + + da9055->regmap = devm_regmap_init_i2c(i2c, &da9055_regmap_config); + if (IS_ERR(da9055->regmap)) { + ret = PTR_ERR(da9055->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da9055, &da9055_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + ret); + } + return ret; +} + +static int __devexit da9055_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da9055_i2c_id[] = { + { "da9055", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da9055_i2c_driver = { + .driver = { + .name = "da9055", + .owner = THIS_MODULE, + }, + .probe = da9055_i2c_probe, + .remove = __devexit_p(da9055_remove), + .id_table = da9055_i2c_id, +}; + +module_i2c_driver(da9055_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); +MODULE_AUTHOR("David Chen, Ashish Chavan"); +MODULE_LICENSE("GPL"); -- 1.7.1

2 3

[alsa-devel] Audio codec volume control
by Rajeev kumar 20 Sep '12

20 Sep '12

Hello Mark, We have three volume control in audio codec. 1. Master volume: control both left and right channel volume 2. left volume: control left channel vol 3. right volume: control right channel vol. So we can provide interfaces for all volume control to the user. Please find below the control I can provide to the user space case 1: 1. Provide master volume control 2. Provide Left channel volume control. 3. Provide Right channel volume control. case 2: 1. Provide master volume control 2. Instead of providing separate control for left and right channel, combined it and provide a single control to user space. so that left and right volume is always in sync. Please let me know which case is better and why? Thank you very much for your support. Best Regards Rajeev

2 5

[alsa-devel] [PATCH] ALSA: usb - disable broken hw volume for Tenx TP6911
by David Henningsson 20 Sep '12

20 Sep '12

While going through Ubuntu bugs, I discovered this patch being posted and a confirmation that the patch works as expected. Finding out how the hw volume really works would be preferrable to just disabling the broken one, but this would be better than nothing. Credit: sndfnsdfin (qawsnews) BugLink: https://bugs.launchpad.net/bugs/559939 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/mixer.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f40ba8..fe56c9d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1267,6 +1267,13 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void /* disable non-functional volume control */ master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME); break; + case USB_ID(0x1130, 0xf211): + snd_printk(KERN_INFO + "usbmixer: volume control quirk for Tenx TP6911 Audio Headset\n"); + /* disable non-functional volume control */ + channels = 0; + break; + } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(bmaControls + csize, csize); -- 1.7.9.5

2 1

Re: [alsa-devel] snd_usb changes broke audio
by Daniel Mack 20 Sep '12

20 Sep '12

(Cc the ALSA list - please don't drop it in replies) On 20.09.2012 02:16, Nate Garner wrote: > I recently updated my kernel from 3.5.3 to 3.5.4. Upon rebooting, I no > longer got any audio through my external sound card, but my headphones > worked. I noticed in the kernel change log that you made some changes to > snd-usb, and thought that might be the culprit. I hope you're the right > person to report this too, if not, feel free to ignore. > Relevant info: > Distro: arch linux, x86_64 > sound card: creative labs model SB0490 (soundblaser Live! 24-bit > external (link here > <http://support.creative.com/Products/ProductDetails.aspx?catID=1&subCatID=2…>) > alsa version 1.0.26 > pulseaudio version 2.1 Argh! I believed we finally fixed all the regressions. Are there any messages in the logs or in dmesg that look releated? Can you send the output of "lsusb -v", please? Daniel

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[alsa-devel] Internal speaker output from Sony Vaio Z (2010 model) lost somewhere between 3.1.0 and 3.2.6
by Adam Williamson 20 Sep '12

20 Sep '12

Welp, pretty much as the topic says. I just noticed today that sound is no longer playing from the internal speakers of my laptop, a 2010 model Sony Vaio Z. Plugging in headphones causes sound to be played fine through those. There are three possible choices for 'Connector' - 'Analog Speakers', 'Analog Output', and 'Analog Headphones' - but none of these seems to make sound come out of the internal speakers. The default is 'Analog Speakers', and when it's set to this, headphone output does work when headphones are plugged in. I've verified that it's broken on Fedora 16 kernels 3.2.6-4, 3.2.8-3 and 3.2.10-1. It's also broken in a very recent Fedora 17 kernel, 3.3.0rc6 or so. It works on a Fedora 16 live image, with kernel 3.1.0-1. Unfortunately the kernels before 3.2.6 have been trashed from Fedora's buildsystem archives, I think, so it's hard to narrow things down any further :/ the alsa-info output is http://www.alsa-project.org/db/?f=368e2359757b537d4daa0f5399830d3942540196 . Thanks! -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Twitter: AdamW_Fedora | identi.ca: adamwfedora http://www.happyassassin.net

3 22

[alsa-devel] [PATCH 00/34] i.MX multi-platform support
by Shawn Guo 20 Sep '12

20 Sep '12

The series enables multi-platform support for imx. Since the required frameworks (clk, pwm) and spare_irq have already been adopted on imx, the series is all about cleaning up mach/* headers. Along with the changes, arch/arm/plat-mxc gets merged into arch/arm/mach-imx. It's based on a bunch of branches (works from others), Rob's initial multi-platform series, Arnd's platform-data and smp_ops (Marc's) and imx 3.7 material (Sascha and myself). It's available on branch below. git://git.linaro.org/people/shawnguo/linux-2.6.git imx/multi-platform It's been tested on imx5 and imx6, and only compile-tested on imx2 and imx3, so testing on imx2/3 are appreciated. Subsystem maintainers, I plan to send the whole series via arm-soc tree at the end of 3.7 merge window when all dependant bits hit mainline. Please have a look at the patches you get copied and provide ACKs if the changes are good. Thanks. Shawn Guo (34): ARM: imx: include board headers in the same folder ASoC: mx27vis: retrieve gpio numbers from platform_data ARM: imx: move iomux drivers and headers into mach-imx ARM: imx: remove unnecessary inclusion from device-imx*.h ARM: imx: move platform device code into mach-imx ARM: imx: merge plat-mxc into mach-imx ARM: imx: include common.h rather than mach/common.h ARM: imx: ARM: imx: include cpuidle.h rather than mach/cpuidle.h ARM: imx: include iim.h rather than mach/iim.h ARM: imx: include iram.h rather than mach/iram.h ARM: imx: include ulpi.h rather than mach/ulpi.h media: mx1_camera: remove the driver ARM: imx: remove mach/dma-mx1-mx2.h dma: ipu: rename mach/ipu.h to include/linux/dma/ipu-dma.h dma: imx-sdma: remove unneeded mach/hardware.h inclusion ASoC: imx-ssi: remove unneeded mach/hardware.h inclusion usb: ehci-mxc: remove unneeded mach/hardware.h inclusion video: mx3fb: remove unneeded mach/hardware.h inclusion watchdog: imx2_wdt: remove unneeded mach/hardware.h inclusion i2c: imx: remove mach/hardware.h inclusion mtd: mxc_nand: remove mach/hardware.h inclusion rtc: mxc_rtc: remove mach/hardware.h inclusion dma: imx-dma: use devm_kzalloc and devm_request_irq dma: imx-dma: retrieve MEM and IRQ from resources dma: imx-dma: remove mach/hardware.h inclusion media: mx2_camera: remove dead code in mx2_camera_add_device media: mx2_camera: use managed functions to clean up code media: mx2_camera: remove mach/hardware.h inclusion mmc: mxcmmc: remove mach/hardware.h inclusion video: imxfb: remove mach/hardware.h inclusion ARM: imx: move debug macros to include/debug ARM: imx: include hardware.h rather than mach/hardware.h ARM: imx: remove header file mach/irqs.h ARM: imx: enable multi-platform build .../devicetree/bindings/i2c/fsl-imx-i2c.txt | 4 +- arch/arm/Kconfig | 15 +- arch/arm/Kconfig.debug | 8 + arch/arm/Makefile | 1 - arch/arm/boot/dts/imx27.dtsi | 4 +- arch/arm/boot/dts/imx51.dtsi | 4 +- arch/arm/boot/dts/imx53.dtsi | 6 +- arch/arm/boot/dts/imx6q.dtsi | 6 +- .../mach/debug-macro.S => include/debug/imx.S} | 33 +- arch/arm/{plat-mxc => mach-imx}/3ds_debugboard.c | 2 +- .../include/mach => mach-imx}/3ds_debugboard.h | 0 arch/arm/mach-imx/Kconfig | 86 ++ arch/arm/mach-imx/Makefile | 23 +- arch/arm/{plat-mxc => mach-imx}/avic.c | 5 +- .../include/mach => mach-imx}/board-mx31lilly.h | 0 .../include/mach => mach-imx}/board-mx31lite.h | 0 .../include/mach => mach-imx}/board-mx31moboard.h | 0 .../include/mach => mach-imx}/board-pcm038.h | 0 arch/arm/mach-imx/clk-imx1.c | 15 +- arch/arm/mach-imx/clk-imx21.c | 14 +- arch/arm/mach-imx/clk-imx25.c | 26 +- arch/arm/mach-imx/clk-imx27.c | 36 +- arch/arm/mach-imx/clk-imx31.c | 21 +- arch/arm/mach-imx/clk-imx35.c | 13 +- arch/arm/mach-imx/clk-imx51-imx53.c | 15 +- arch/arm/mach-imx/clk-imx6q.c | 3 +- arch/arm/mach-imx/clk-pllv1.c | 4 +- .../{plat-mxc/include/mach => mach-imx}/common.h | 0 arch/arm/mach-imx/cpu-imx25.c | 5 +- arch/arm/mach-imx/cpu-imx27.c | 2 +- arch/arm/mach-imx/cpu-imx31.c | 7 +- arch/arm/mach-imx/cpu-imx35.c | 5 +- arch/arm/mach-imx/cpu-imx5.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpu.c | 3 +- arch/arm/mach-imx/cpu_op-mx51.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpufreq.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpuidle.c | 0 .../{plat-mxc/include/mach => mach-imx}/cpuidle.h | 0 arch/arm/mach-imx/devices-imx1.h | 3 +- arch/arm/mach-imx/devices-imx21.h | 3 +- arch/arm/mach-imx/devices-imx25.h | 3 +- arch/arm/mach-imx/devices-imx27.h | 3 +- arch/arm/mach-imx/devices-imx31.h | 3 +- arch/arm/mach-imx/devices-imx35.h | 3 +- arch/arm/mach-imx/devices-imx50.h | 3 +- arch/arm/mach-imx/devices-imx51.h | 3 +- arch/arm/{plat-mxc => mach-imx}/devices/Kconfig | 3 - arch/arm/{plat-mxc => mach-imx}/devices/Makefile | 3 +- .../mach => mach-imx/devices}/devices-common.h | 19 +- arch/arm/{plat-mxc => mach-imx/devices}/devices.c | 1 - .../devices/platform-ahci-imx.c | 5 +- .../{plat-mxc => mach-imx}/devices/platform-fec.c | 5 +- .../devices/platform-flexcan.c | 4 +- .../devices/platform-fsl-usb2-udc.c | 5 +- .../devices/platform-gpio-mxc.c | 2 +- .../devices/platform-gpio_keys.c | 5 +- .../devices/platform-imx-dma.c | 23 +- .../devices/platform-imx-fb.c | 16 +- .../devices/platform-imx-i2c.c | 31 +- .../devices/platform-imx-keypad.c | 4 +- .../devices/platform-imx-ssi.c | 4 +- .../devices/platform-imx-uart.c | 4 +- .../devices/platform-imx2-wdt.c | 5 +- .../devices/platform-imx21-hcd.c | 4 +- .../devices/platform-imx_udc.c | 4 +- .../devices/platform-imxdi_rtc.c | 5 +- .../devices/platform-ipu-core.c | 5 +- .../devices/platform-mx2-camera.c | 16 +- .../devices/platform-mxc-ehci.c | 5 +- .../devices/platform-mxc-mmc.c | 20 +- .../devices/platform-mxc_nand.c | 25 +- .../devices/platform-mxc_pwm.c | 4 +- .../devices/platform-mxc_rnga.c | 4 +- .../devices/platform-mxc_rtc.c | 13 +- .../devices/platform-mxc_w1.c | 4 +- .../devices/platform-pata_imx.c | 4 +- .../devices/platform-sdhci-esdhc-imx.c | 5 +- .../devices/platform-spi_imx.c | 4 +- arch/arm/mach-imx/ehci-imx25.c | 4 +- arch/arm/mach-imx/ehci-imx27.c | 4 +- arch/arm/mach-imx/ehci-imx31.c | 4 +- arch/arm/mach-imx/ehci-imx35.c | 4 +- arch/arm/mach-imx/ehci-imx5.c | 4 +- arch/arm/{plat-mxc => mach-imx}/epit.c | 6 +- .../include/mach => mach-imx}/eukrea-baseboards.h | 0 arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 7 +- arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 8 +- arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 7 +- arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 7 +- .../{plat-mxc/include/mach => mach-imx}/hardware.h | 26 +- arch/arm/mach-imx/hotplug.c | 3 +- arch/arm/{plat-mxc/include/mach => mach-imx}/iim.h | 0 arch/arm/mach-imx/imx27-dt.c | 11 +- arch/arm/mach-imx/imx31-dt.c | 5 +- arch/arm/mach-imx/imx51-dt.c | 9 +- arch/arm/mach-imx/include/mach/dma-mx1-mx2.h | 10 - arch/arm/mach-imx/iomux-imx31.c | 5 +- .../include/mach => mach-imx}/iomux-mx1.h | 2 +- .../include/mach => mach-imx}/iomux-mx21.h | 4 +- .../include/mach => mach-imx}/iomux-mx25.h | 2 +- .../include/mach => mach-imx}/iomux-mx27.h | 4 +- .../include/mach => mach-imx}/iomux-mx2x.h | 0 .../include/mach => mach-imx}/iomux-mx3.h | 0 .../include/mach => mach-imx}/iomux-mx35.h | 2 +- .../include/mach => mach-imx}/iomux-mx50.h | 2 +- .../include/mach => mach-imx}/iomux-mx51.h | 2 +- arch/arm/{plat-mxc => mach-imx}/iomux-v1.c | 5 +- .../{plat-mxc/include/mach => mach-imx}/iomux-v1.h | 0 arch/arm/{plat-mxc => mach-imx}/iomux-v3.c | 5 +- .../{plat-mxc/include/mach => mach-imx}/iomux-v3.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/iram.h | 0 arch/arm/{plat-mxc => mach-imx}/iram_alloc.c | 3 +- arch/arm/{plat-mxc => mach-imx}/irq-common.c | 0 arch/arm/{plat-mxc => mach-imx}/irq-common.h | 3 + arch/arm/mach-imx/lluart.c | 3 +- arch/arm/mach-imx/mach-apf9328.c | 7 +- arch/arm/mach-imx/mach-armadillo5x0.c | 9 +- arch/arm/mach-imx/mach-bug.c | 7 +- arch/arm/mach-imx/mach-cpuimx27.c | 11 +- arch/arm/mach-imx/mach-cpuimx35.c | 9 +- arch/arm/mach-imx/mach-cpuimx51sd.c | 9 +- arch/arm/mach-imx/mach-eukrea_cpuimx25.c | 10 +- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 48 +- arch/arm/mach-imx/mach-imx27ipcam.c | 6 +- arch/arm/mach-imx/mach-imx27lite.c | 6 +- arch/arm/mach-imx/mach-imx53.c | 11 +- arch/arm/mach-imx/mach-imx6q.c | 6 +- arch/arm/mach-imx/mach-kzm_arm11_01.c | 7 +- arch/arm/mach-imx/mach-mx1ads.c | 7 +- arch/arm/mach-imx/mach-mx21ads.c | 6 +- arch/arm/mach-imx/mach-mx25_3ds.c | 8 +- arch/arm/mach-imx/mach-mx27_3ds.c | 10 +- arch/arm/mach-imx/mach-mx27ads.c | 6 +- arch/arm/mach-imx/mach-mx31_3ds.c | 12 +- arch/arm/mach-imx/mach-mx31ads.c | 5 +- arch/arm/mach-imx/mach-mx31lilly.c | 11 +- arch/arm/mach-imx/mach-mx31lite.c | 11 +- arch/arm/mach-imx/mach-mx31moboard.c | 14 +- arch/arm/mach-imx/mach-mx35_3ds.c | 8 +- arch/arm/mach-imx/mach-mx50_rdp.c | 7 +- arch/arm/mach-imx/mach-mx51_3ds.c | 9 +- arch/arm/mach-imx/mach-mx51_babbage.c | 7 +- arch/arm/mach-imx/mach-mxt_td60.c | 6 +- arch/arm/mach-imx/mach-pca100.c | 8 +- arch/arm/mach-imx/mach-pcm037.c | 8 +- arch/arm/mach-imx/mach-pcm037_eet.c | 5 +- arch/arm/mach-imx/mach-pcm038.c | 13 +- arch/arm/mach-imx/mach-pcm043.c | 9 +- arch/arm/mach-imx/mach-qong.c | 6 +- arch/arm/mach-imx/mach-scb9328.c | 7 +- arch/arm/mach-imx/mach-vpr200.c | 7 +- arch/arm/mach-imx/mm-imx1.c | 9 +- arch/arm/mach-imx/mm-imx21.c | 12 +- arch/arm/mach-imx/mm-imx25.c | 10 +- arch/arm/mach-imx/mm-imx27.c | 12 +- arch/arm/mach-imx/mm-imx3.c | 9 +- arch/arm/mach-imx/mm-imx5.c | 8 +- arch/arm/mach-imx/mx1-camera-fiq-ksym.c | 18 - arch/arm/mach-imx/mx1-camera-fiq.S | 35 - arch/arm/{plat-mxc/include/mach => mach-imx}/mx1.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx21.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx25.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx27.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx2x.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx31.h | 0 arch/arm/mach-imx/mx31lilly-db.c | 9 +- arch/arm/mach-imx/mx31lite-db.c | 9 +- arch/arm/mach-imx/mx31moboard-devboard.c | 9 +- arch/arm/mach-imx/mx31moboard-marxbot.c | 9 +- arch/arm/mach-imx/mx31moboard-smartbot.c | 11 +- .../arm/{plat-mxc/include/mach => mach-imx}/mx35.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx3x.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx50.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx51.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx53.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx6q.h | 0 arch/arm/{plat-mxc/include/mach => mach-imx}/mxc.h | 0 arch/arm/mach-imx/pcm970-baseboard.c | 7 +- arch/arm/mach-imx/platsmp.c | 5 +- arch/arm/mach-imx/pm-imx27.c | 3 +- arch/arm/mach-imx/pm-imx3.c | 7 +- arch/arm/mach-imx/pm-imx5.c | 7 +- arch/arm/mach-imx/pm-imx6q.c | 5 +- arch/arm/{plat-mxc => mach-imx}/ssi-fiq-ksym.c | 0 arch/arm/{plat-mxc => mach-imx}/ssi-fiq.S | 0 arch/arm/{plat-mxc => mach-imx}/system.c | 5 +- arch/arm/{plat-mxc => mach-imx}/time.c | 5 +- arch/arm/{plat-mxc => mach-imx}/tzic.c | 6 +- arch/arm/{plat-mxc => mach-imx}/ulpi.c | 2 +- .../arm/{plat-mxc/include/mach => mach-imx}/ulpi.h | 0 arch/arm/plat-mxc/Kconfig | 89 -- arch/arm/plat-mxc/Makefile | 24 - arch/arm/plat-mxc/devices/platform-mx1-camera.c | 42 - arch/arm/plat-mxc/include/mach/irqs.h | 21 - arch/arm/plat-mxc/include/mach/timex.h | 22 - arch/arm/plat-mxc/include/mach/uncompress.h | 132 --- drivers/dma/imx-dma.c | 137 +-- drivers/dma/imx-sdma.c | 1 - drivers/dma/ipu/ipu_idmac.c | 3 +- drivers/dma/ipu/ipu_irq.c | 3 +- drivers/i2c/busses/i2c-imx.c | 40 +- drivers/media/video/Kconfig | 12 - drivers/media/video/Makefile | 1 - drivers/media/video/mx1_camera.c | 889 -------------------- drivers/media/video/mx2_camera.c | 240 +++--- drivers/media/video/mx3_camera.c | 2 +- drivers/mmc/host/mxcmmc.c | 31 +- drivers/mtd/nand/mxc_nand.c | 86 +- drivers/rtc/rtc-mxc.c | 34 +- drivers/usb/host/ehci-mxc.c | 1 - drivers/video/imxfb.c | 38 +- drivers/video/mx3fb.c | 3 +- drivers/watchdog/imx2_wdt.c | 1 - .../mach/ipu.h => include/linux/dma/ipu-dma.h | 6 +- include/linux/platform_data/asoc-mx27vis.h | 11 + include/linux/platform_data/camera-mx1.h | 35 - sound/soc/fsl/imx-pcm-fiq.c | 3 +- sound/soc/fsl/imx-ssi.c | 1 - sound/soc/fsl/mx27vis-aic32x4.c | 42 +- 219 files changed, 1149 insertions(+), 2170 deletions(-) rename arch/arm/{plat-mxc/include/mach/debug-macro.S => include/debug/imx.S} (59%) rename arch/arm/{plat-mxc => mach-imx}/3ds_debugboard.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/3ds_debugboard.h (100%) rename arch/arm/{plat-mxc => mach-imx}/avic.c (98%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31lilly.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31lite.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31moboard.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-pcm038.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/common.h (100%) rename arch/arm/{plat-mxc => mach-imx}/cpu.c (97%) rename arch/arm/{plat-mxc => mach-imx}/cpufreq.c (99%) rename arch/arm/{plat-mxc => mach-imx}/cpuidle.c (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/cpuidle.h (100%) rename arch/arm/{plat-mxc => mach-imx}/devices/Kconfig (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/Makefile (96%) rename arch/arm/{plat-mxc/include/mach => mach-imx/devices}/devices-common.h (96%) rename arch/arm/{plat-mxc => mach-imx/devices}/devices.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-ahci-imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-fec.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-flexcan.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-fsl-usb2-udc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-gpio-mxc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-gpio_keys.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-dma.c (63%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-fb.c (79%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-i2c.c (77%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-keypad.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-ssi.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-uart.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx2-wdt.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx21-hcd.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx_udc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imxdi_rtc.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-ipu-core.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mx2-camera.c (83%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc-ehci.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc-mmc.c (76%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_nand.c (74%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_pwm.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_rnga.c (95%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_rtc.c (77%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_w1.c (95%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-pata_imx.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-sdhci-esdhc-imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-spi_imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/epit.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/eukrea-baseboards.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/hardware.h (94%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iim.h (100%) delete mode 100644 arch/arm/mach-imx/include/mach/dma-mx1-mx2.h rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx1.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx21.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx25.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx27.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx2x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx3.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx35.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx50.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx51.h (99%) rename arch/arm/{plat-mxc => mach-imx}/iomux-v1.c (98%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-v1.h (100%) rename arch/arm/{plat-mxc => mach-imx}/iomux-v3.c (97%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-v3.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iram.h (100%) rename arch/arm/{plat-mxc => mach-imx}/iram_alloc.c (98%) rename arch/arm/{plat-mxc => mach-imx}/irq-common.c (100%) rename arch/arm/{plat-mxc => mach-imx}/irq-common.h (94%) delete mode 100644 arch/arm/mach-imx/mx1-camera-fiq-ksym.c delete mode 100644 arch/arm/mach-imx/mx1-camera-fiq.S rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx1.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx21.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx25.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx27.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx2x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx31.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx35.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx3x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx50.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx51.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx53.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx6q.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mxc.h (100%) rename arch/arm/{plat-mxc => mach-imx}/ssi-fiq-ksym.c (100%) rename arch/arm/{plat-mxc => mach-imx}/ssi-fiq.S (100%) rename arch/arm/{plat-mxc => mach-imx}/system.c (97%) rename arch/arm/{plat-mxc => mach-imx}/time.c (99%) rename arch/arm/{plat-mxc => mach-imx}/tzic.c (98%) rename arch/arm/{plat-mxc => mach-imx}/ulpi.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/ulpi.h (100%) delete mode 100644 arch/arm/plat-mxc/Kconfig delete mode 100644 arch/arm/plat-mxc/Makefile delete mode 100644 arch/arm/plat-mxc/devices/platform-mx1-camera.c delete mode 100644 arch/arm/plat-mxc/include/mach/irqs.h delete mode 100644 arch/arm/plat-mxc/include/mach/timex.h delete mode 100644 arch/arm/plat-mxc/include/mach/uncompress.h delete mode 100644 drivers/media/video/mx1_camera.c rename arch/arm/plat-mxc/include/mach/ipu.h => include/linux/dma/ipu-dma.h (97%) create mode 100644 include/linux/platform_data/asoc-mx27vis.h delete mode 100644 include/linux/platform_data/camera-mx1.h -- 1.7.9.5

4 20

[alsa-devel] [PATCH v2 1/7] ASoC: fsl: add PPC_MPC52xx dependency to SND_POWERPC_SOC
by "Eric Millbrandt 20 Sep '12

20 Sep '12

From: Eric Millbrandt <emillbrandt(a)dekaresearch.com> mpc52xx socs need to define SND_POWERPC_SOC to build ASoC drivers. Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- Changes since v1: Patch was "powerpc/52xx: define FSL_SOC" Changed from defining FSL_SOC for PPC_MPC52xx to adding PPC_MPC52xx as a dependency of SND_POWERPC_SOC as per Anatolij Gustschin's comment. sound/soc/fsl/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d701330..4563b28 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -6,7 +6,7 @@ config SND_SOC_FSL_UTILS menuconfig SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" - depends on FSL_SOC + depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. -- 1.7.2.5

3 3

[alsa-devel] [RFC PATCH v2 0/5] snd-ice1712: Add Philips PSC724 Ultimate Edge
by Ondrej Zary 20 Sep '12

20 Sep '12

Hello, this is second attempt to add Philips PSC724 Ultimate Edge sound card support to snd-ice1712 driver. Unlike other snd-ice1712 subdrivers, the codec code is splitted into separate modules and thus reusable by other ice1712 subdrivers. Working: all analog outputs (front, rear, center+lfe, headphone) and inputs (front mic, rear mic, line, cd, aux), mixer, headphone jack detection, suspend/resume Currently untested: SPDIF Buggy: switching AGC mode in mixer causes alsamixer to crash - looks like it does not like notifications about control activating/deactivating

2 9

[alsa-devel] [RFC 0/7] mpc5200 ASoC and pcm030 board fixes
by Eric Millbrandt 19 Sep '12

19 Sep '12

This series is a respin of "mpc5200 ASoC fixups" The mpc5200 ASoC and pcm030 board code compiled, but did not run after the multi-codec patches. This series add the missing pieces into the mpc5200 ASoC drivers and updates the pcm030 board to the current api. Eric Millbrandt (7): powerpc/52xx: define FSL_SOC ASoC: fsl: mpc5200 combine psc_dma platform data ASoC: fsl: mpc5200 add missing information to snd_soc_dai_driver ASoC: fsl: cleanup headers in pcm030-audio-fabric ASoC: fsl: convert pcm030-audio-fabric to a platform-driver ASoC: fsl: convert pcm030-audio-fabric to use snd_soc_register_card ASoC: fsl: register the wm9712-codec arch/powerpc/platforms/52xx/Kconfig | 1 + sound/soc/fsl/mpc5200_dma.c | 24 ++------- sound/soc/fsl/mpc5200_dma.h | 3 + sound/soc/fsl/mpc5200_psc_ac97.c | 10 ++++ sound/soc/fsl/mpc5200_psc_i2s.c | 8 +++ sound/soc/fsl/pcm030-audio-fabric.c | 100 +++++++++++++++++++++++++--------- 6 files changed, 99 insertions(+), 47 deletions(-) -- 1.7.2.5

2 13

[alsa-devel] [PATCH 1/5] [v3] ASoC: fsl: use snd_soc_register_card to register the card
by Timur Tabi 19 Sep '12

19 Sep '12

Use snd_soc_register_card() instead of platform_device_alloc("soc-audio") to register the sound card from the machine drivers. The use of platform_device_alloc is deprecated. Although several other drivers still use platform_device_alloc(), the Freescale drivers were not using it to pass driver data. Instead of fixing the driver data usage, it's better to replace the deprecated code. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 32 +++++++------------------------- sound/soc/fsl/p1022_ds.c | 31 ++++++------------------------- 2 files changed, 13 insertions(+), 50 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 60bcba1..9ff9318 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -192,7 +192,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; int ret = -ENODEV; const char *sprop; @@ -341,34 +340,22 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->card.probe = mpc8610_hpcd_machine_probe; machine_data->card.remove = mpc8610_hpcd_machine_remove; machine_data->card.name = pdev->name; /* The platform driver name */ + machine_data->card.owner = THIS_MODULE; + machine_data->card.dev = &pdev->dev; machine_data->card.num_links = 2; machine_data->card.dai_link = machine_data->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &machine_data->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&machine_data->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error_sound; + dev_err(&pdev->dev, "could not register card\n"); + goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; -error_sound: - platform_device_put(sound_device); error: kfree(machine_data); error_alloc: @@ -383,17 +370,12 @@ error_alloc: */ static int __devexit mpc8610_hpcd_remove(struct platform_device *pdev) { - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - platform_device_unregister(sound_device); - + snd_soc_unregister_card(card); kfree(machine_data); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); return 0; } diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 50adf40..144d496 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -202,7 +202,6 @@ static int p1022_ds_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct machine_data *mdata; int ret = -ENODEV; const char *sprop; @@ -349,36 +348,23 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->card.probe = p1022_ds_machine_probe; mdata->card.remove = p1022_ds_machine_remove; mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.owner = THIS_MODULE; + mdata->card.dev = &pdev->dev; mdata->card.num_links = 2; mdata->card.dai_link = mdata->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &mdata->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&mdata->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; error: - if (sound_device) - platform_device_put(sound_device); - kfree(mdata); error_put: of_node_put(codec_np); @@ -392,17 +378,12 @@ error_put: */ static int __devexit p1022_ds_remove(struct platform_device *pdev) { - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); struct machine_data *mdata = container_of(card, struct machine_data, card); - platform_device_unregister(sound_device); - + snd_soc_unregister_card(card); kfree(mdata); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.7.3.4

3 14

Re: [alsa-devel] [PATCH 07/22] ASoC: Ux500: Initialise PCM from MSP probe rather than as a device
by Lee Jones 19 Sep '12

19 Sep '12

Mark, > > > I'm sorry but this patch is breaking the design of ASoC. > > > The ASoC-platform is the DMA-block (in combination with the > > > MSP-block), and there should be a platform-driver for the DMA/PCM. > > > The > > > platform-driver then has a DAI which is the MSP. The ASoC > > > DAI-link-struct should have one driver for each of these, so the > > > dummy-driver for PCM should be there. > > > > Just thinking about this now. I converted it to the current format at > > the request of Mark. If this isn't the correct method I'm not quite > > sure what is. If you want it to be registered as a device, then it > > needs to go into the Device Tree, but Mark doesn't want it in there > > because it doesn't actually represent hardware. I've just taken a closer look at this with a view to finding the most suitable solution. My conclusion is that although the PCM doesn't contain any registers, or represent hardware it should be a device and therefore be present in the Device Tree. These are my findings: PCM devices already represented in DTs: fsl,mpc5200-pcm - written by Grant Likely, the author of Device Tree phytec,pcm030 - written by Grant Likely, the author of Device Tree PCM devices which register as actual devices (should be in DT): samsung - samsung-pcm sh - siu-pcm-audio omap - omap-pcm-audio pxa - pxa-pcm-audio jz4740 - jz4740-pcm-audio kirkwood - kirkwood-pcm-audio ep93xx - ep93xx-pcm-audio ... The later was basically every PCM device bar one I think. The Open Firmware used to stipulate that each device represented in the Device Tree to own registers and therefore be an actual hardware device, but that has since been lifted as it didn't make sense. I propose to represent the PCM in the Device Tree again and have it probe just like all the other PCM devices in sound/soc. If you give me the nod, I'll revert this patch, enable the PCM for DT again and resent the patch-set in full. -- Lee Jones Linaro ST-Ericsson Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

2 8

[alsa-devel] [Resend PATCH] ASoC: wm8904: remove redundant code
by Bo Shen 19 Sep '12

19 Sep '12

The core_intercon register two times for wm8904 Remove one Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index dc4262e..7c8df52 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1185,8 +1185,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets, ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, core_intercon, - ARRAY_SIZE(core_intercon)); snd_soc_dapm_add_routes(dapm, adc_intercon, ARRAY_SIZE(adc_intercon)); snd_soc_dapm_add_routes(dapm, dac_intercon, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - avoid non-standard "Docking" name in mixers
by David Henningsson 19 Sep '12

19 Sep '12

The standard name (and what PulseAudio picks up) is "Dock Mic", not "Docking Mic" or "Docking-Station". Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_analog.c | 14 +++++++------- sound/pci/hda/patch_conexant.c | 4 ++-- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a82cce..cdd43ea 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1810,7 +1810,7 @@ static const struct hda_input_mux ad1981_hp_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, - { "Docking-Station", 0x1 }, + { "Dock Mic", 0x1 }, { "Mix", 0x2 }, }, }; @@ -1834,8 +1834,8 @@ static const struct snd_kcontrol_new ad1981_hp_mixers[] = { */ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Docking-Station Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Docking-Station Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), /* FIXME: does this laptop have analog CD connection? */ @@ -3565,7 +3565,7 @@ static const struct hda_amp_list ad1884_loopbacks[] = { static const char * const ad1884_slave_vols[] = { "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", - "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958", + "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958", NULL }; @@ -3626,7 +3626,7 @@ static const struct hda_input_mux ad1984_thinkpad_capture_source = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, { "Mix", 0x3 }, - { "Docking-Station", 0x4 }, + { "Dock Mic", 0x4 }, }, }; @@ -3655,8 +3655,8 @@ static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7d70210..b871b013 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1710,8 +1710,8 @@ static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT), {} }; -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH v4 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 19 Sep '12

19 Sep '12

Hi Samuel, On 09/19/2012 03:24 AM, Samuel Ortiz wrote: > Hi Peter, > > On Mon, Sep 10, 2012 at 01:46:18PM +0300, Peter Ujfalusi wrote: >> Hello, >> >> Generated on top of: >> git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap > I applied the first 8 patches, but I'd like to get Mark's ACK for the rest of > the serie. > Unless you're expecting Mark to take the whole thing, in which case you can > add my: > > Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> > > to the MFD patches. > Please let me know how you'd like to proceed. If Mark is still OK with it I think it would be better to queue this series via ASoC so we can keep them in one place (also I think patch 14 needs patch 6 in order to not break x86/x86_64 allmodconfig). On the other hand if they go separately (1-8 via MFD, 9-14 via ASoC) they will meet in linux-next where this matters the most. Thank you, Péter

1 0

[alsa-devel] Need help with WM8960
by Timur Tabi 19 Sep '12

19 Sep '12

Mark, I'm trying to port some old Linux code to upstream for a third-party board that uses a Freescale P1022 and a Wolfson WM8960. I've got audio working, but it's using a hack from the third-party code, and I don't know how to do implement the change properly, mostly because I don't understand DAPM. The first problem is that I get this: wm8960 0-001a: No platform data supplied What kind of platform data is the driver expecting, and how do I get it that data? I don't seem to need any platform data to make the code work. Second, I uploaded the hack to wm8960.c at http://pastebin.com/RwVy6cQg You can see that he disables the DAPM controls for DACL, DACR, ADCL, and ADCR, and forces those bits set in the wm8960_probe function. There's also a comment that documents some limitations. What is the proper way to implement this change? Third, in order to actually hear audio, I had to unmute the "Left Output Mixer PCM" and "Right Output Mixer PCM". Is this something that the machine driver should do automatically? And if so, how? -- Timur Tabi Linux kernel developer at Freescale

3 7

[alsa-devel] [PATCH] ASoC: imx-audmux: remove null check of audmux_base in audmux_read_file
by Richard Zhao 19 Sep '12

19 Sep '12

When audmux_read_file is called, it means the driver is already initialised successfully, so we don't need to check audmux_base. It also fix smatch warning: sound/soc/fsl/imx-audmux.c:78 audmux_read_file() warn: possible memory leak of 'buf' Signed-off-by: Richard Zhao <richard.zhao(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index e7c800e..524ce62 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -74,9 +74,6 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (!audmux_base) - return -ENOSYS; - if (audmux_clk) clk_prepare_enable(audmux_clk); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: imx-pcm-dma: check kzalloc return value in function snd_imx_open
by Richard Zhao 19 Sep '12

19 Sep '12

It fixed smatch warning: sound/soc/fsl/imx-pcm-dma.c:112 snd_imx_open() error: potential null dereference 'dma_data'. (kzalloc returns null) Signed-off-by: Richard Zhao <richard.zhao(a)freescale.com> --- sound/soc/fsl/imx-pcm-dma.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 48f9d88..a23505a 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -109,6 +109,9 @@ static int snd_imx_open(struct snd_pcm_substream *substream) dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); + if (!dma_data) + return -ENOMEM; + dma_data->peripheral_type = dma_params->shared_peripheral ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: Avoid recalculating the bitmask for SOC_ENUM controls
by Lars-Peter Clausen 19 Sep '12

19 Sep '12

For ENUM controls the bitmask is calculated based on the number of items. Currently this is done each time the control is accessed. And while the performance impact of this should be negligible we can easily do better. The roundup_pow_of_two macro performs the same calculation which is currently done manually, but it is also possible to use this macro with compile time constants and so it can be used to initialize static data. So we can use it to initialize the mask field of a ENUM control during its declaration. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 +++- sound/soc/codecs/twl4030.c | 8 +++----- sound/soc/soc-core.c | 16 ++++++---------- sound/soc/soc-dapm.c | 22 ++++++++-------------- 4 files changed, 20 insertions(+), 30 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 313b766..91244a0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -20,6 +20,7 @@ #include <linux/interrupt.h> #include <linux/kernel.h> #include <linux/regmap.h> +#include <linux/log2.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/compress_driver.h> @@ -160,7 +161,8 @@ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .max = xmax, .texts = xtexts } + .max = xmax, .texts = xtexts, \ + .mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0} #define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts) #define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 391fcfc..2548f5c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -999,7 +999,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val; - unsigned short mask, bitmask; + unsigned short mask; if (twl4030->configured) { dev_err(codec->dev, @@ -1007,18 +1007,16 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, return -EBUSY; } - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits(codec, e->reg, mask, val); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e5b0713..9a6daf9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2413,16 +2413,14 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] - = (val >> e->shift_l) & (bitmask - 1); + = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2443,19 +2441,17 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; - unsigned int mask, bitmask; + unsigned int mask; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits_locked(codec, e->reg, mask, val); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3b8ddb4..873e6e7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -358,12 +358,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mux: { struct soc_enum *e = (struct soc_enum *) w->kcontrol_news[i].private_value; - int val, item, bitmask; + int val, item; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = soc_widget_read(w, e->reg); - item = (val >> e->shift_l) & (bitmask - 1); + item = (val >> e->shift_l) & e->mask; p->connect = 0; for (i = 0; i < e->max; i++) { @@ -2680,15 +2678,13 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(widget->codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); + ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2712,22 +2708,20 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; - unsigned int mask, bitmask; + unsigned int mask; struct snd_soc_dapm_update update; int wi; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); -- 1.7.10.4

3 2

[alsa-devel] [PATCH v2 0/3] Allow suspend/resume with USB audio.
by Dylan Reid 18 Sep '12

18 Sep '12

Problem being addressed: shell one$ aplay -Dhw:<usbcard> file.wav shell two$ echo mem > /sys/power/state After resume, aplay attempts resume, then restart, then after a timeout gets a write error and no audio will be played until it is restarted. During suspend/resume, if the USB port remains powered the audio device will still be present upon resume. The interface and endpoint have to be reconfigured however. These patches move the configuration from hw_params to prepare causing re-configuration during resume. Is this the right way to fix this? Or is there an easier way I am missing? Thanks for looking. v2: fix unused variable warnings. Dylan Reid (3): ALSA: usb-audio: set period_bytes in substream. ALSA: usb-audio: Don't require hw_params in endpoint. ALSA: usb-audio: Move configuration to prepare. sound/usb/card.h | 2 + sound/usb/endpoint.c | 29 +++++++----- sound/usb/endpoint.h | 5 ++- sound/usb/pcm.c | 121 +++++++++++++++++++++++++++++++-------------------- 4 files changed, 97 insertions(+), 60 deletions(-) -- 1.7.12.146.g16d26b1

2 5

[alsa-devel] [PATCH 0/3] Allow suspend/resume with USB audio.
by Dylan Reid 18 Sep '12

18 Sep '12

Problem being addressed: shell one$ aplay -Dhw:<usbcard> file.wav shell two$ echo mem > /sys/power/state After resume, aplay attempts resume, then restart, then after a timeout gets a write error and no audio will be played until it is restarted. During suspend/resume, if the USB port remains powered the audio device will still be present upon resume. The interface and endpoint have to be reconfigured however. These patches move the configuration from hw_params to prepare causing re-configuration during resume. Is this the right way to fix this? Or is there an easier way I am missing? Thanks for looking. Dylan Reid (3): ALSA: usb-audio: set period_bytes in substream. ALSA: usb-audio: Don't require hw_params in endpoint. ALSA: usb-audio: Move configuration to prepare. sound/usb/card.h | 2 + sound/usb/endpoint.c | 29 +++++++----- sound/usb/endpoint.h | 5 ++- sound/usb/pcm.c | 121 +++++++++++++++++++++++++++++++-------------------- 4 files changed, 97 insertions(+), 60 deletions(-) -- 1.7.12.146.g16d26b1

2 8

[alsa-devel] what is wrong?
by Eva Lovi 18 Sep '12

18 Sep '12

Hello, Does Alsa support this soundcard? card 0: I82801AAICH [Intel 82801AA-ICH], device 0: Intel ICH [Intel 82801AA-ICH] I tried aplay, but I don't hear anything. In amixer I increased the volume, but I still don't hear anything. What's the problem? Eva Lovi

1 1

[alsa-devel] [PATCH] ALSA: hda - limit internal mic boost for Asus X202E
by David Henningsson 18 Sep '12

18 Sep '12

When the input gain for the internal mic is set to its maximum level, the background noise becomes so high - and any relevant signal clipped - that the setting becomes unusable. It is better to limit the amplification. BugLink: https://bugs.launchpad.net/bugs/1052460 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Feel free to add cc to stable if you think it makes sense to do so. Alsa info available at: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1052460/+attachment/33… sound/pci/hda/patch_via.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5f36a1..74fb6fd 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3666,6 +3666,32 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) update_power_state(codec, 0x21, AC_PWRST_D3); } +/* + * pin fix-up + */ +enum { + VIA_FIXUP_INTMIC_BOOST, +}; + +static void via_fixup_intmic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + override_mic_boost(codec, 0x30, 0, 2, 40); +} + +static const struct hda_fixup via_fixups[] = { + [VIA_FIXUP_INTMIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = via_fixup_intmic_boost, + }, +}; + +static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), + {} +}; + /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) { @@ -3682,6 +3708,9 @@ static int patch_vt2002P(struct hda_codec *codec) override_mic_boost(codec, 0x29, 0, 3, 40); add_secret_dac_path(codec); + snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) { -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 00/15] ASoC: OMAP: Convert to use dmaengine
by Peter Ujfalusi 18 Sep '12

18 Sep '12

Hello, Changelog: - Support for pause/resume for OMAP audio via dmaengine - dmaengine: support for NO_PERIOD_WAKEUP in cyclic mode - OMAP to keep supporting NO_PERIOD_WAKEUP for audio - Other plaforms can also try to enable this mode since we have now generic interface to do so. This series will switch the OMAP audio to use dmaengine. The final patch which does the switch was based on Russell King's earlier patch. The first 10 patch is to prepare the OMAP audio drivers for a smooth change to dmaengine: - sDMA FRAME sync mode is removed and replaced with PACKET mode - dai drivers no longer need to configure sDMA sync mode - dai drivers does not need to specify the DMA word length - with the exception of the omap-hdmi driver which requires 32bit word length regardless of the audio format in use - the McPDM driver used (to my surprise) hackish way of getting the DMA channel and address - via defines from some header files After the conversion OMAP audio support should have the same features as before, no regressions expected. I have tested the series on: - BeagleBoard (audio via McBSP): - aplay/arecord. In element mode and in threshold mode with different period sizes - mplayer -ao alsa: for direct ALSA access - mplayer -ao pulse: via PulseAudio to test NO_PERIOD_WAKEUP feature - OMAP4 Blaze (audio via McPDM and DMIC) - aplay/arecord - mplayer -ao alsa: for direct ALSA access - mplayer -ao pulse: via PulseAudio to test NO_PERIOD_WAKEUP feature The patches has been generated against: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 Janusz: Can you retest this series on OMAP1 to be sure I have not broken it? Ricardo: Can you test the omap-hmdi if it is still working? Regards, Peter --- Peter Ujfalusi (15): dmaengine: omap: Support for element mode in cyclic DMA dmaengine: omap: Add support for pause/resume in cyclic dma mode dmaengine: Add no_wakeup parameter to dmaengine_prep_dma_cyclic() dmaengine: Pass no_wakeup parameter via device_prep_dma_cyclic() callback dmaengine: omap-dma: Add support for no_wakeup in cyclic mode ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode ASoC: omap-pcm: Select sDMA synchronization based on packet_size ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges ASoC: omap-mcpdm: Use platform_get_resource_* to get resources ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type ASoC: omap-pcm, omap-dmic: Change the use of omap_pcm_dma_data->data_type ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time ASoC: omap-pcm: Convert to use dmaengine arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + drivers/dma/at_hdmac.c | 3 +- drivers/dma/ep93xx_dma.c | 4 +- drivers/dma/imx-dma.c | 2 +- drivers/dma/imx-sdma.c | 2 +- drivers/dma/mmp_tdma.c | 2 +- drivers/dma/mxs-dma.c | 2 +- drivers/dma/omap-dma.c | 47 ++++-- drivers/dma/pl330.c | 2 +- drivers/dma/sa11x0-dma.c | 2 +- drivers/dma/sirf-dma.c | 2 +- drivers/dma/ste_dma40.c | 3 +- drivers/dma/tegra20-apb-dma.c | 2 +- include/linux/dmaengine.h | 8 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 17 ++- sound/soc/omap/omap-mcbsp.c | 60 +++----- sound/soc/omap/omap-mcpdm.c | 40 +++-- sound/soc/omap/omap-pcm.c | 236 ++++++++--------------------- sound/soc/omap/omap-pcm.h | 4 +- sound/soc/soc-dmaengine-pcm.c | 3 +- 22 files changed, 186 insertions(+), 269 deletions(-) -- 1.7.12

7 36

[alsa-devel] Channel map API patches: to be 3.7 or not to be?
by Takashi Iwai 17 Sep '12

17 Sep '12

Hi, does anyone have concern if I push the current channel map API patches (in sound-unstable git tree topic/tlv-chmap branch) for linux-next, i.e. for inclusion to 3.7 kernel? I don't worry too much about the kernel API. This can be refactored later. But the kernel ABI must retain, so we won't change in an incompatible way any longer once after it's merged to the upstream. So, if anyone sees a flaw in the kernel ABI definition using control elements, let me know. (But note that the API can't be perfect for all generic purposes. It's designed to be "good enough" for a big lack for multi-channel streaming. Of course, this doesn't discourage to use the API for any other purposes.) thanks, Takashi

3 12

[alsa-devel] [PATCH] hda-emu: Add support for channel mapping API
by David Henningsson 17 Sep '12

17 Sep '12

--- Hi Takashi, I tried to make hda-emu compile with the latest changes to the HDA driver. This is the result. It compiles, but when running any codec, I get the error "Control element Playback Channel Map:0 already exists!". I'm unsure whether this is a problem in the driver or in my simple copy-paste attempt to make this work. Can you offer some insight? Thanks. include/sound/asound.h | 57 +++++++++++++ include/sound/pcm.h | 58 +++++++++++++ include/sound/tlv.h | 8 ++ snd-wrapper.c | 214 ++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 337 insertions(+) diff --git a/include/sound/asound.h b/include/sound/asound.h index 25e90ed..61f1440 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -164,3 +164,60 @@ struct snd_ctl_event { unsigned char data8[60]; } data; }; + +typedef int __bitwise snd_pcm_state_t; +#define SNDRV_PCM_STATE_OPEN ((snd_pcm_state_t) 0) /* stream is open */ +#define SNDRV_PCM_STATE_SETUP ((snd_pcm_state_t) 1) /* stream has a setup */ +#define SNDRV_PCM_STATE_PREPARED ((snd_pcm_state_t) 2) /* stream is ready to start */ +#define SNDRV_PCM_STATE_RUNNING ((snd_pcm_state_t) 3) /* stream is running */ +#define SNDRV_PCM_STATE_XRUN ((snd_pcm_state_t) 4) /* stream reached an xrun */ +#define SNDRV_PCM_STATE_DRAINING ((snd_pcm_state_t) 5) /* stream is draining */ +#define SNDRV_PCM_STATE_PAUSED ((snd_pcm_state_t) 6) /* stream is paused */ +#define SNDRV_PCM_STATE_SUSPENDED ((snd_pcm_state_t) 7) /* hardware is suspended */ +#define SNDRV_PCM_STATE_DISCONNECTED ((snd_pcm_state_t) 8) /* hardware is disconnected */ +#define SNDRV_PCM_STATE_LAST SNDRV_PCM_STATE_DISCONNECTED + +struct snd_pcm_mmap_status { + snd_pcm_state_t state; /* RO: state - SNDRV_PCM_STATE_XXXX */ +}; + + +/* channel positions */ +enum { + SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_MONO, /* mono stream */ + /* this follows the alsa-lib mixer channel value + 3 */ + SNDRV_CHMAP_FL, /* front left */ + SNDRV_CHMAP_FR, /* front right */ + SNDRV_CHMAP_RL, /* rear left */ + SNDRV_CHMAP_RR, /* rear right */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_LFE, /* LFE */ + SNDRV_CHMAP_SL, /* side left */ + SNDRV_CHMAP_SR, /* side right */ + SNDRV_CHMAP_RC, /* rear center */ + /* new definitions */ + SNDRV_CHMAP_FLC, /* front left center */ + SNDRV_CHMAP_FRC, /* front right center */ + SNDRV_CHMAP_RLC, /* rear left center */ + SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_FLW, /* front left wide */ + SNDRV_CHMAP_FRW, /* front right wide */ + SNDRV_CHMAP_FLH, /* front left high */ + SNDRV_CHMAP_FCH, /* front center high */ + SNDRV_CHMAP_FRH, /* front right high */ + SNDRV_CHMAP_TC, /* top center */ + SNDRV_CHMAP_TFL, /* top front left */ + SNDRV_CHMAP_TFR, /* top front right */ + SNDRV_CHMAP_TFC, /* top front center */ + SNDRV_CHMAP_TRL, /* top rear left */ + SNDRV_CHMAP_TRR, /* top rear right */ + SNDRV_CHMAP_TRC, /* top rear center */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, +}; + +#define SNDRV_CHMAP_POSITION_MASK 0xffff +#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) +#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) + diff --git a/include/sound/pcm.h b/include/sound/pcm.h index f1719cb..4980c69 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -45,6 +45,8 @@ struct snd_pcm_runtime { unsigned int info; unsigned int rate_num; unsigned int rate_den; + /* -- mmap -- */ + struct snd_pcm_mmap_status *status; void *private_data; void (*private_free)(struct snd_pcm_runtime *runtime); @@ -62,12 +64,21 @@ struct snd_pcm_substream { struct snd_pcm_ops *ops; /* -- runtime information -- */ struct snd_pcm_runtime *runtime; + /* -- next substream -- */ + struct snd_pcm_substream *next; void *file; int ref_count; + }; struct snd_pcm_str { + /* -- substreams -- */ unsigned int substream_opened; + unsigned int substream_count; + struct snd_pcm_substream *substream; + + struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ + }; struct snd_pcm { @@ -241,5 +252,52 @@ static inline int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *run, int snd_pcm_format_width(int fmt); +/* + * PCM channel-mapping control API + */ +/* array element of channel maps */ +struct snd_pcm_chmap_elem { + unsigned char channels; + unsigned char map[15]; +}; + +/* channel map information; retrieved via snd_kcontrol_chip() */ +struct snd_pcm_chmap { + struct snd_pcm *pcm; /* assigned PCM instance */ + int stream; /* PLAYBACK or CAPTURE */ + struct snd_kcontrol *kctl; + const struct snd_pcm_chmap_elem *chmap; + unsigned int max_channels; + unsigned int channel_mask; /* optional: active channels bitmask */ + void *private_data; /* optional: private data pointer */ +}; + +/* get the PCM substream assigned to the given chmap info */ +static inline struct snd_pcm_substream * +snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx) +{ + struct snd_pcm_substream *s; + for (s = info->pcm->streams[info->stream].substream; s; s = s->next) + if (s->number == idx) + return s; + return NULL; +} + +/* ALSA-standard channel maps (RL/RR prior to C/LFE) */ +extern const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[]; +/* Other world's standard channel maps (C/LFE prior to RL/RR) */ +extern const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[]; + +/* bit masks to be passed to snd_pcm_chmap.channel_mask field */ +#define SND_PCM_CHMAP_MASK_24 ((1U << 2) | (1U << 4)) +#define SND_PCM_CHMAP_MASK_246 (SND_PCM_CHMAP_MASK_24 | (1U << 6)) +#define SND_PCM_CHMAP_MASK_2468 (SND_PCM_CHMAP_MASK_246 | (1U << 8)) + +int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_chmap_elem *chmap, + int max_channels, + unsigned long private_value, + struct snd_pcm_chmap **info_ret); + #endif /* __SOUND_PCM_H */ diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 7067e2d..de36aaa 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -73,4 +73,12 @@ #define TLV_DB_GAIN_MUTE -9999999 +/* + * channel-mapping TLV items + * TLV length must match with num_channels + */ +#define SNDRV_CTL_TLVT_CHMAP_FIXED 0x101 /* fixed channel position */ +#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */ +#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */ + #endif /* __SOUND_TLV_H */ diff --git a/snd-wrapper.c b/snd-wrapper.c index aaa2467..7716e67 100644 --- a/snd-wrapper.c +++ b/snd-wrapper.c @@ -257,3 +257,217 @@ void mylock_unlock(int *lock, const char *file, int line) break; } } + +/* + * standard channel mapping helpers + */ +#include <sound/tlv.h> + +/* default channel maps for multi-channel playbacks, up to 8 channels */ +const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_MONO } }, + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { .channels = 8, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; +EXPORT_SYMBOL_GPL(snd_pcm_std_chmaps); + +/* alternative channel maps with CLFE <-> surround swapped for 6/8 channels */ +const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_MONO } }, + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 8, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; +EXPORT_SYMBOL_GPL(snd_pcm_alt_chmaps); + +static bool valid_chmap_channels(const struct snd_pcm_chmap *info, int ch) +{ + if (ch > info->max_channels) + return false; + return !info->channel_mask || (info->channel_mask & (1U << ch)); +} + +static int pcm_chmap_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 0; + uinfo->count = info->max_channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SNDRV_CHMAP_LAST; + return 0; +} + +/* get callback for channel map ctl element + * stores the channel position firstly matching with the current channels + */ +static int pcm_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct snd_pcm_substream *substream; + const struct snd_pcm_chmap_elem *map; + + if (snd_BUG_ON(!info->chmap)) + return -EINVAL; + substream = snd_pcm_chmap_substream(info, idx); + if (!substream) + return -ENODEV; + memset(ucontrol->value.integer.value, 0, + sizeof(ucontrol->value.integer.value)); + if (!substream->runtime) + return 0; /* no channels set */ + for (map = info->chmap; map->channels; map++) { + int i; + if (map->channels == substream->runtime->channels && + valid_chmap_channels(info, map->channels)) { + for (i = 0; i < map->channels; i++) + ucontrol->value.integer.value[i] = map->map[i]; + return 0; + } + } + return -EINVAL; +} + +/* tlv callback for channel map ctl element + * expands the pre-defined channel maps in a form of TLV + */ +static int pcm_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + const struct snd_pcm_chmap_elem *map; + unsigned int __user *dst; + int c, count = 0; + + if (snd_BUG_ON(!info->chmap)) + return -EINVAL; + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CONTAINER, tlv)) + return -EFAULT; + size -= 8; + dst = tlv + 2; + for (map = info->chmap; map->channels; map++) { + int chs_bytes = map->channels * 4; + if (!valid_chmap_channels(info, map->channels)) + continue; + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CHMAP_FIXED, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + size -= chs_bytes; + count += chs_bytes; + for (c = 0; c < map->channels; c++) { + if (put_user(map->map[c], dst)) + return -EFAULT; + dst++; + } + } + if (put_user(count, tlv + 1)) + return -EFAULT; + return 0; +} + +static void pcm_chmap_ctl_private_free(struct snd_kcontrol *kcontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + info->pcm->streams[info->stream].chmap_kctl = NULL; + kfree(info); +} + +/** + * snd_pcm_add_chmap_ctls - create channel-mapping control elements + * @pcm: the assigned PCM instance + * @stream: stream direction + * @chmap: channel map elements (for query) + * @max_channels: the max number of channels for the stream + * @private_value: the value passed to each kcontrol's private_value field + * @info_ret: store struct snd_pcm_chmap instance if non-NULL + * + * Create channel-mapping control elements assigned to the given PCM stream(s). + * Returns zero if succeed, or a negative error value. + */ +int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_chmap_elem *chmap, + int max_channels, + unsigned long private_value, + struct snd_pcm_chmap **info_ret) +{ + struct snd_pcm_chmap *info; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .info = pcm_chmap_ctl_info, + .get = pcm_chmap_ctl_get, + .tlv.c = pcm_chmap_ctl_tlv, + }; + int err; + + info = kzalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + info->pcm = pcm; + info->stream = stream; + info->chmap = chmap; + info->max_channels = max_channels; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + knew.name = "Playback Channel Map"; + else + knew.name = "Capture Channel Map"; + knew.device = pcm->device; + knew.count = pcm->streams[stream].substream_count; + knew.private_value = private_value; + info->kctl = snd_ctl_new1(&knew, info); + if (!info->kctl) { + kfree(info); + return -ENOMEM; + } + info->kctl->private_free = pcm_chmap_ctl_private_free; + err = snd_ctl_add(pcm->card, info->kctl); + if (err < 0) + return err; + pcm->streams[stream].chmap_kctl = info->kctl; + if (info_ret) + *info_ret = info; + return 0; +} +EXPORT_SYMBOL_GPL(snd_pcm_add_chmap_ctls); -- 1.7.9.5

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[alsa-devel] period events in sw_params
by Prasant J 17 Sep '12

17 Sep '12

Hi, What is period event? (when used in snd_pcm_sw_params_set_period_event for playback) -Prasant

1 0

[alsa-devel] sw_params configuration clarification
by Prasant J 17 Sep '12

17 Sep '12

Hi, I'm using ALSA API (alsa-lib 1.0.25) for writing an audio application for an embedded product. This is my understanding of the sw_params configuration api: (Can someone please confirm if the explanation is correct?) snd_pcm_sw_params_set_start_threshold -- sets the minimum no. of frames before which playback will not start. snd_pcm_sw_params_stop_threshold -- sets the minimum no. of frames when the playback stops (not completely clear) snd_pcm_sw_params_set_silence_threshold -- sets the no. of frames for inserting silence (equal to set by set_silence_size) when frames is less than the value set. Any info will be of help. -Prasant

1 0

[alsa-devel] [PATCH RESEND 1/5] ARM: tegra: config: enable dmaengine based APB DMA driver
by Stephen Warren 17 Sep '12

17 Sep '12

From: Laxman Dewangan <ldewangan(a)nvidia.com> Enable config for dmaengine based Tegra APB DMA driver and disable the legacy APB DMA driver (SYSTEM_DMA). Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- I'm looking for Ack's on patches 4 and 5 from the SPI and ASoC maintainers respectively, so that I can take this series through the Tegra tree for 3.7. When this series was first posted, there was a bug in the Tegra ASoC PCM driver which prevented it from working. A fix for this has since been applied to ASoC's for-3.6 branch and hence should show up in 3.6. arch/arm/configs/tegra_defconfig | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/arm/configs/tegra_defconfig b/arch/arm/configs/tegra_defconfig index adcf3c2..2990396 100644 --- a/arch/arm/configs/tegra_defconfig +++ b/arch/arm/configs/tegra_defconfig @@ -29,6 +29,7 @@ CONFIG_ARCH_TEGRA_2x_SOC=y CONFIG_ARCH_TEGRA_3x_SOC=y CONFIG_TEGRA_PCI=y CONFIG_TEGRA_DEBUG_UART_AUTO_ODMDATA=y +# CONFIG_TEGRA_SYSTEM_DMA is not set CONFIG_TEGRA_EMC_SCALING_ENABLE=y CONFIG_SMP=y CONFIG_PREEMPT=y @@ -175,6 +176,8 @@ CONFIG_RTC_DRV_MAX8907=y CONFIG_RTC_DRV_TPS65910=y CONFIG_RTC_DRV_EM3027=y CONFIG_RTC_DRV_TEGRA=y +CONFIG_DMADEVICES=y +CONFIG_TEGRA20_APB_DMA=y CONFIG_STAGING=y CONFIG_SENSORS_ISL29018=y CONFIG_SENSORS_ISL29028=y -- 1.7.0.4

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Re: [alsa-devel] [PATCH] ALSA: hda - Remove ignore_misc_bit
by Raymond Yau 17 Sep '12

17 Sep '12

>>>>>> But I still wonder why PulseAudio cares the headphone jack state even >>>>>> though this has only one output at all? >>>>> >>>>> >>>>> When seeing the system as a whole, there can be other outputs on other >>>>> cards - HDMI, USB etc. If somebody e g plugs a USB headset in it will be >>>>> simpler for the user if PulseAudio does not also show the unplugged 3.5 >>>>> mm jack phantom port of internal speaker or internal Mic is ok but how about those jack spdif, CD , Aux which does not have jack detect circuit Is it feasible for driver implement a read/ write kcontrol for those port and allow the user to change the state of those port by using special application (e.g. qashctl) ? > > That said, it's not super difficult to comment out a few lines in > /usr/share/pulseaudio/alsa-mixer/paths/*.conf, and also, most mixer UIs (e g pavucontrol) still allows you to route audio to an unavailable port. For ad1988 which still not support auto mute nor hda jack detect. The user still able to use the card in gnome but not in unity. https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/995169 And ca0132 which does not implement auto mute ? https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1043621 And the other ad198x which support auto mute but not implement hda jack detect https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1051494 The other cases are those on-board ac97 controller (e.g. Intel8x0, atiixp,...) with ac97 codecs which have the headphone playback volume control. What is the usage of those "Headphone Jack Sense" control of AC97 codecs ? Some of them seem have inverted control https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1033996

1 0

[alsa-devel] [PATCH] ALSA: hda - Add another pci id for Haswell board
by Wang Xingchao 17 Sep '12

17 Sep '12

A new PCI id 0x0d0c for Haswell HDA Controller. [root@SKBM04SDP ~]# lspci |grep Audio 00:03.0 Audio device: Intel Corporation Device 0d0c (rev 02) 00:1b.0 Audio device: Intel Corporation Lynx Point HD Audio Controller Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c9c779..e54673d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3442,6 +3442,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + { PCI_DEVICE(0x8086, 0x0d0c), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: compress - move the buffer check
by Vinod Koul 17 Sep '12

17 Sep '12

Commit ALSA: compress_core: integer overflow in snd_compr_allocate_buffer() added a new error check for input params. this add new routine for input checks and moves buffer overflow check to this new routine. This allows the error value to be propogated to user space Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/core/compress_offload.c | 20 ++++++++++++++++---- 1 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 68fe02c..bd7f28e 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -407,10 +407,6 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, unsigned int buffer_size; void *buffer; - if (params->buffer.fragment_size == 0 || - params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) - return -EINVAL; - buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { buffer = NULL; @@ -429,6 +425,16 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, return 0; } +static int snd_compress_check_input(struct snd_compr_params *params) +{ + /* first let's check the buffer parameter's */ + if (params->buffer.fragment_size == 0 || + params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + return -EINVAL; + + return 0; +} + static int snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) { @@ -447,11 +453,17 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) retval = -EFAULT; goto out; } + + retval = snd_compress_check_input(params); + if (retval) + goto out; + retval = snd_compr_allocate_buffer(stream, params); if (retval) { retval = -ENOMEM; goto out; } + retval = stream->ops->set_params(stream, params); if (retval) goto out; -- 1.7.0.4

2 3

[alsa-devel] ucm: hardware resource arbitration
by Patrick Lai 16 Sep '12

16 Sep '12

Hi, I have two use cases which can acquire same CODEC path but they cannot run concurrently through mixing. When such scenario does arise, arbitration is required based on priority of use cases. On the other hand, same use cases on different platform may use different CODEC paths so arbitration is not required. In order to decide whether arbitration is required, I think the best place to do so is in UCM. I looked at existing UCM APIs but I do not believe existing APIs have concept of arbitration built in. If I am wrong, how can I handle arbitration with existing UCM APIs. If I am correct, is there anyone looking to expand UCM APIs for similar scenario? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 3

[alsa-devel] ASoC updates for 3.6
by Mark Brown 15 Sep '12

15 Sep '12

The following changes since commit d9875690d9b89a866022ff49e3fcea892345ad92: Linux 3.6-rc2 (2012-08-16 14:51:24 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 985b11fa8064d55d0d5a84e68667434598911bb2: ASoC: wm8904: correct the index (2012-09-14 18:02:38 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.6 A bigger set of updates than I'm entirely comfortable with - things backed up a bit due to travel. As ever the majority of these are small, focused updates for specific drivers though there are a couple of core changes. There's been good exposure in -next. The AT91 patch fixes a build break. ---------------------------------------------------------------- Bo Shen (1): ASoC: wm8904: correct the index Dylan Reid (1): ASoC: samsung dma - Don't indicate support for pause/resume. Fabio Estevam (1): ASoC: mc13783: Remove mono support Heather Lomond (1): ASoC: arizona: Fix typo in 44.1kHz rates Joachim Eastwood (1): ASoC: atmel-ssc: include linux/io.h for raw io Julia Lawall (3): ASoC: imx-sgtl5000: fix error return code ASoC: ux500_msp_i2s: better use devm functions and fix error return code ASoC: am3517evm: fix error return code Mark Brown (2): ASoC: dapm: Make sure we update the bias level for CODECs with no op ASoC: dapm: Don't force card bias level to be updated Prasad Joshi (1): ASoC: spear: correct the check for NULL dma_buffer pointer Stephen Warren (2): sound: tegra_alc5632: remove HP detect GPIO inversion ASoC: tegra: fix maxburst settings in dmaengine code include/linux/atmel-ssc.h | 1 + sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/mc13783.c | 8 ++++---- sound/soc/codecs/wm8904.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/omap/am3517evm.c | 2 +- sound/soc/samsung/dma.c | 8 +------- sound/soc/soc-dapm.c | 5 ++++- sound/soc/spear/spear_pcm.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 1 - sound/soc/tegra/tegra_pcm.c | 4 ++-- sound/soc/ux500/ux500_msp_i2s.c | 25 +++++-------------------- 12 files changed, 22 insertions(+), 40 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: wm8904: correct the index
by Bo Shen 14 Sep '12

14 Sep '12

Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 65d525d..13e8222 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = { { 14, 0x0000 }, /* R14 - Power Management 2 */ { 15, 0x0000 }, /* R15 - Power Management 3 */ { 18, 0x0000 }, /* R18 - Power Management 6 */ - { 19, 0x945E }, /* R20 - Clock Rates 0 */ + { 20, 0x945E }, /* R20 - Clock Rates 0 */ { 21, 0x0C05 }, /* R21 - Clock Rates 1 */ { 22, 0x0006 }, /* R22 - Clock Rates 2 */ { 24, 0x0050 }, /* R24 - Audio Interface 0 */ -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 5/5] ASoC: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the I2S FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to I2S FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Please note that I've applied this to Tegra's for-3.7/dmaengine branch, even though it hasn't been ack'd by an ASoC maintainer.

2 3

Re: [alsa-devel] [PATCH 5/5] ASoC: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the I2S FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to I2S FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Mark, Liam, I'd like to take this patch through the Tegra tree, since it relies on the previous Tegra patches to convert to dmaengine. Could I please get an ack if that's OK? Thanks.

2 2

Re: [alsa-devel] [PATCH 4/7] mfd: ab8500-core: Apply the AB8500 CODEC's compatible string to its MFD cell
by Mark Brown 14 Sep '12

14 Sep '12

On Fri, Jul 27, 2012 at 03:36:33PM +0200, Samuel Ortiz wrote: > On Fri, Jul 27, 2012 at 01:38:51PM +0100, Lee Jones wrote: > > Provide a compatible string for the AB8500 CODEC to aid in > > configuration property look-up from its associated Device Tree > > node. > Applied as well. I remain *deeply* unconvinced that this is tasteful design for the device tree bindings, and it's certainly not needed urgently for v3.6 as there's no DT bindings on the CODEC side.

3 2

Re: [alsa-devel] [PATCH 4/5] spi: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the SPI FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to SPI FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Grant, Mark, I'd like to take this patch through the Tegra tree, since it relies on the previous Tegra patches to convert to dmaengine. Could I please get an ack if that's OK? Thanks.

2 3

[alsa-devel] [PATCH] ASoC: wm8960: Support shared LRCLK
by Mark Brown 14 Sep '12

14 Sep '12

If the LRCLK is shared and the WM8960 is clock master then we should enable the LRCM bit to tell the device that it should drive LRCLK when either ADC or DAC is enabled rather than separately driving the two LRCLKs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm8960.h | 2 ++ sound/soc/codecs/wm8960.c | 11 +++++++++++ 2 files changed, 13 insertions(+) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index 74e9a95..b5a1ab9e 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -19,6 +19,8 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ int dres; /* Discharge resistance for headphone outputs */ + + bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ }; #endif diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 7cb0d07c..8740423 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1029,6 +1029,7 @@ static const struct regmap_config wm8960_regmap = { static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; @@ -1041,6 +1042,16 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata && pdata->shared_lrclk) { + ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, + 0x4, 0x4); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", + ret); + return ret; + } + } + i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.10.4

2 3

Re: [alsa-devel] [PATCH 1/7] powerpc/52xx: define FSL_SOC
by Eric Millbrandt 14 Sep '12

14 Sep '12

On 2012-09-14 agust(a)denx.de wrote: > Eric Millbrandt <emillbrandt(a)dekaresearch.com> wrote: > ... > > Better use: > > diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index > d701330..4563b28 100644 > --- a/sound/soc/fsl/Kconfig > +++ b/sound/soc/fsl/Kconfig > @@ -6,7 +6,7 @@ config SND_SOC_FSL_UTILS > > menuconfig SND_POWERPC_SOC > tristate "SoC Audio for Freescale PowerPC CPUs" > - depends on FSL_SOC > + depends on FSL_SOC || PPC_MPC52xx > help > Say Y or M if you want to add support for codecs attached to > the PowerPC CPUs. > Thanks, > > Anatolij > Looks good to me, I will do this instead. Thanks. Eric -DISCLAIMER: an automatically appended disclaimer may follow. By posting- -to a public e-mail mailing list I hereby grant permission to distribute- -and copy this message.- This e-mail and the information, including any attachments, it contains are intended to be a confidential communication only to the person or entity to whom it is addressed and may contain information that is privileged. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please immediately notify the sender and destroy the original message. Thank you. Please consider the environment before printing this email.

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Re: [alsa-devel] FW: [PATCH 07/22] ASoC: Ux500: Initialise PCM from MSP probe rather than as a device
by Ola Lilja 14 Sep '12

14 Sep '12

On 08/23/2012 09:29 PM, Ola Lilja wrote: > > >> -----Original Message----- >> From: Lee Jones [mailto:lee.jones@linaro.org] >> Sent: den 23 augusti 2012 16:59 >> To: Mark Brown >> Cc: Ola Lilja; 'Linus Walleij'; roger.xr.nilsson(a)stericsson.com; linux- >> arm-kernel(a)lists.infradead.org; alsa-devel(a)alsa-project.org >> Subject: Re: [PATCH 07/22] ASoC: Ux500: Initialise PCM from MSP probe >> rather than as a device >> >> On Thu, Aug 23, 2012 at 03:37:57PM +0100, Mark Brown wrote: >> > On Thu, Aug 23, 2012 at 02:26:19PM +0100, Lee Jones wrote: >> > >> > > > I think Ola is suggesting probing the DMA driver from the machine >> > > > which will also work though I'm not 100% sure if I'm parsing the >> > > > above correctly. The issue in DT terms is that if the DMA >> > > > controller is shared with a bunch of other IPs then it should >> have >> > > > one node shared between them all and not a bunch of shim nodes >> > > > inserted in the middle which only exists due to the way Linux >> instantiates stuff. I haven't followed everything in this discussion, but what I meant in an earlier mail was that using the device for platform-DAI (MSP_I2S) also as platform int the DAI-link struct is not anything I can see fit (as in the patch Lee first submitted). I can see a few acceptable solutions: 1) Use it as it is! (A virtual device for PCM is created in arch/arm and the probe is invoked from the machine-driver). 2) Since the actual platform is the DMA-block, maybe there is some way of making the device "non-virtual" by having some sub-device in the DMA-driver that is instead triggered in the same way as in 1). 3) Move it to DT, but it seems that it is not possible when it is now a virtual device. Any of these three would be fine with me. >> > >> > > When you say 'machine', do you mean from arch/<arch>/mach-*? If so, >> > > I'm keen for that not to happen. >> > >> > No, sound/soc/ux500/snowball.c or whatever. At least that's my >> guess. >> >> Ah, I see. Maybe the mop500.c file then. >> >> > > > They instantiate the PCM driver dynamically from the DAI when >> it's >> > > > probed which is pretty much what you're patch is doing. >> > >> > > So they do it in the same why I have with this patch? Do you know >> > > why Ola might think this is a bad idea? >> > >> > I'm not 100% sure, I'm guessing it might be down to the fact that you >> > end up with multiple PCM drivers. We could avoid that with >> > refcounting but nobody's really worried about it. >> >> I think I'll wait for Ola to get back, as he's the expert on this >> stuff. >> >> I'll attempt to re-jig the patch-set, as this is a blocker atm. >> >> -- >> Lee Jones >> Linaro ST-Ericsson Landing Team Lead >> Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook >> | Twitter | Blog >

3 2

[alsa-devel] [PATCH 00/17] MOP500 Audio fix-ups and DT enablement
by Lee Jones 14 Sep '12

14 Sep '12

This patch-set provides the MOP500 Audio driver with a few small fix-ups along with Device Tree enablement for MOP500 Core and the MSPs. You'll notice that we've left the PCM out for the time being and continue to register that using platform code. This is because we're still in the middle of discussions pertaining to how this pseudo-device should be handled. We also pass pinctrl responsibility to the MSP driver instead of setting it up in platform code, which is an unnecessary over-head. Documentation/devicetree/bindings/mfd/ab8500.txt | 15 ++ .../devicetree/bindings/sound/ux500-mop500.txt | 39 +++ .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++ arch/arm/boot/dts/dbx5x0.dtsi | 39 +++ arch/arm/boot/dts/snowball.dts | 16 ++ arch/arm/mach-ux500/Makefile | 2 +- arch/arm/mach-ux500/board-mop500-audio.c | 199 +++++++++++++++ arch/arm/mach-ux500/board-mop500-msp.c | 267 -------------------- arch/arm/mach-ux500/board-mop500-msp.h | 14 - arch/arm/mach-ux500/board-mop500.c | 22 +- arch/arm/mach-ux500/board-mop500.h | 8 + arch/arm/mach-ux500/include/mach/msp.h | 2 - include/linux/mfd/abx500/ab8500-codec.h | 6 +- sound/soc/codecs/ab8500-codec.c | 81 ++++++ sound/soc/ux500/mop500.c | 47 +++- sound/soc/ux500/ux500_msp_dai.c | 6 + sound/soc/ux500/ux500_msp_i2s.c | 89 +++++-- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 18 files changed, 584 insertions(+), 319 deletions(-)

5 35

[alsa-devel] [patch] ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()
by Dan Carpenter 14 Sep '12

14 Sep '12

These are 32 bit values that come from the user, we need to check for integer overflows or we could end up allocating a smaller buffer than expected. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d..5a733e7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -409,6 +409,10 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, unsigned int buffer_size; void *buffer; + if (params->buffer.fragment_size == 0 || + params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + return -EINVAL; + buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { buffer = NULL;

3 10

[alsa-devel] [PATCH 0/8] ALSA: Add DSP firmware loader (resub)
by Ian Minett 14 Sep '12

14 Sep '12

From: Ian Minett <ian_minett(a)creativelabs.com> >Thanks for the patches, and sorry for the late response, No problem, and sorry for ours too- we've been working on fixing the issues you raised last time. Hopefully this submission will sort out many of those items. Macros have been taken out, comments added, and firmware is now cached. >Keep the git commit author as me. Ok. I ensured you were set as the git commit author for the first two patchfiles. Is there anything else I need to do to properly retain your authorship on those commits? >- Why hda_stream_format is defined? hda_stream_format is defined to store the stream format information used for the DSP download. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's patch: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's code: Pass DMA buffer pointers in calls to setup_bdle(). Add DSP loader callback routines to controller. Add new DSP loader switch to Kconfig to enable DSP firmware loading. 3: - patch_ca0132.c - ca0132_regs.h Add DSP register definitions header file 4: - patch_ca0132.c Add DSP firmware enums and defs to CA0132 codec 5: - patch_ca0132.c Add calls to new DSP loader system to transfer firmware binary to the hardware. Add chip read/write routines, DSP I/O, SCP packet format helper functions and transfer DMA management. 6: - patch_ca0132.c Add DSP firmware caching to CA0132 codec 7: - patch_ca0132.c Add comments and descriptions to functions. Merge chipio write address functions and add fix to dsp_write_wait(). 8: - hda_codec.h Change return value for load_dsp_prepare to -ENOSYS in case where DSP loader routines are not available.

2 15

Re: [alsa-devel] [PATCH 4/5] spi: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the SPI FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to SPI FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Please note that I've applied this to Tegra's for-3.7/dmaengine branch, even though it hasn't been ack'd by an SPI maintainer.

1 0

Re: [alsa-devel] [PATCH 0/5] ARM: tegra: move all APB DMA client to dmaengine based driver
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > This patch series enable the dmaengine based Tegra APB DMA driver > and remove the support code for the legacy dma driver from different > APB DMA clients. > Patch 1 enable the dmaengine based dma drie and disable the legacy dma driver > in config file. > Patch 2 remove the legacy dma driver. > Patch 3,4, 5 remove the support code from different APB DMA clients to support > legacy dma driver. > > Please note that this series will depends on patch > dma: tegra: enable/disable dma clock > which is not in 3.6-rc1 but will be available in next RC (rc2) of 3.6 as > part of fixes. I have applied this series to Tegra's for-3.7/dmaengine branch. Right now, that branch is based directly on Mark Brown's ASoC for-3.6 branch in order to pick up the fix it depends on. However, I'll rebase on the 3.6 pull request tag he sends for that branch as soon as it's available.

1 0

Re: [alsa-devel] [PATCH v2] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug unconditionally
by Takashi Iwai 13 Sep '12

13 Sep '12

At Thu, 13 Sep 2012 16:18:08 +0200, Daniel Vetter wrote: > > On Thu, Sep 13, 2012 at 4:00 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > But, this makes me wonder why this patch can't go into 3.6. It's a > > clear bug and the fix is trivial. > > > > Daniel, isn't it better to pick the fix (for 3.6) to drm-intel-fixes, > > then merge back to drm-intel-next? > > Well, on a quick read-through it didn't sound serious (just a bit of > noise in dmesg and a few spurious reconfigs), For the graphics side, yes. But for the sound driver side, you'll miss the jack-detection unplug event without this fix. > and we're rather late in > the -rc cycle. Yes, but a bug is a bug is a bug :) And the fix is two-liners. If there were no conflict, would you mind it? I don't think so. > Applying this on top of -fixes will cause a conflict, so didn't looked > like I should bother. Yeah, that's bad. > Also, I _want_ the patch for -next so that git > blame shows the commit and not some random conflicting merge. With parallel development, you can't always get straight lines, as you know well... Takashi

1 0

[alsa-devel] [PATCH v2] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug unconditionally
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_enable_hdmi(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_disable_hdmi(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 5d02aad..229897f 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -674,10 +674,7 @@ static void intel_disable_hdmi(struct intel_encoder *encoder) struct drm_i915_private *dev_priv = dev->dev_private; struct intel_hdmi *intel_hdmi = enc_to_intel_hdmi(&encoder->base); u32 temp; - u32 enable_bits = SDVO_ENABLE; - - if (intel_hdmi->has_audio) - enable_bits |= SDVO_AUDIO_ENABLE; + u32 enable_bits = SDVO_ENABLE | SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: codecs: Add DA9055 codec driver
by Ashish Chavan 13 Sep '12

13 Sep '12

This patch adds support for Dialog semiconductor's DA9055 audio codec. This has been tested on DA9055 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9055.c | 1507 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1513 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3684255..b92759a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DA732X if I2C + select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC @@ -239,6 +240,9 @@ config SND_SOC_DA7210 config SND_SOC_DA732X tristate +config SND_SOC_DA9055 + tristate + config SND_SOC_DFBMCS320 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ca508b2..9bd4d95 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da732x-objs := da732x.o +snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o @@ -144,6 +145,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o +obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c new file mode 100644 index 0000000..be5c9e6 --- /dev/null +++ b/sound/soc/codecs/da9055.c @@ -0,0 +1,1507 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +/* DA9055 register space */ + +/* Status Registers */ +#define DA9055_STATUS1 0x02 +#define DA9055_PLL_STATUS 0x03 +#define DA9055_AUX_L_GAIN_STATUS 0x04 +#define DA9055_AUX_R_GAIN_STATUS 0x05 +#define DA9055_MIC_L_GAIN_STATUS 0x06 +#define DA9055_MIC_R_GAIN_STATUS 0x07 +#define DA9055_MIXIN_L_GAIN_STATUS 0x08 +#define DA9055_MIXIN_R_GAIN_STATUS 0x09 +#define DA9055_ADC_L_GAIN_STATUS 0x0A +#define DA9055_ADC_R_GAIN_STATUS 0x0B +#define DA9055_DAC_L_GAIN_STATUS 0x0C +#define DA9055_DAC_R_GAIN_STATUS 0x0D +#define DA9055_HP_L_GAIN_STATUS 0x0E +#define DA9055_HP_R_GAIN_STATUS 0x0F +#define DA9055_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA9055_CIF_CTRL 0x20 +#define DA9055_DIG_ROUTING_AIF 0X21 +#define DA9055_SR 0x22 +#define DA9055_REFERENCES 0x23 +#define DA9055_PLL_FRAC_TOP 0x24 +#define DA9055_PLL_FRAC_BOT 0x25 +#define DA9055_PLL_INTEGER 0x26 +#define DA9055_PLL_CTRL 0x27 +#define DA9055_AIF_CLK_MODE 0x28 +#define DA9055_AIF_CTRL 0x29 +#define DA9055_DIG_ROUTING_DAC 0x2A +#define DA9055_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA9055_AUX_L_GAIN 0x30 +#define DA9055_AUX_R_GAIN 0x31 +#define DA9055_MIXIN_L_SELECT 0x32 +#define DA9055_MIXIN_R_SELECT 0x33 +#define DA9055_MIXIN_L_GAIN 0x34 +#define DA9055_MIXIN_R_GAIN 0x35 +#define DA9055_ADC_L_GAIN 0x36 +#define DA9055_ADC_R_GAIN 0x37 +#define DA9055_ADC_FILTERS1 0x38 +#define DA9055_MIC_L_GAIN 0x39 +#define DA9055_MIC_R_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA9055_DAC_FILTERS5 0x40 +#define DA9055_DAC_FILTERS2 0x41 +#define DA9055_DAC_FILTERS3 0x42 +#define DA9055_DAC_FILTERS4 0x43 +#define DA9055_DAC_FILTERS1 0x44 +#define DA9055_DAC_L_GAIN 0x45 +#define DA9055_DAC_R_GAIN 0x46 +#define DA9055_CP_CTRL 0x47 +#define DA9055_HP_L_GAIN 0x48 +#define DA9055_HP_R_GAIN 0x49 +#define DA9055_LINE_GAIN 0x4A +#define DA9055_MIXOUT_L_SELECT 0x4B +#define DA9055_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA9055_SYSTEM_MODES_INPUT 0x50 +#define DA9055_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA9055_AUX_L_CTRL 0x60 +#define DA9055_AUX_R_CTRL 0x61 +#define DA9055_MIC_BIAS_CTRL 0x62 +#define DA9055_MIC_L_CTRL 0x63 +#define DA9055_MIC_R_CTRL 0x64 +#define DA9055_MIXIN_L_CTRL 0x65 +#define DA9055_MIXIN_R_CTRL 0x66 +#define DA9055_ADC_L_CTRL 0x67 +#define DA9055_ADC_R_CTRL 0x68 +#define DA9055_DAC_L_CTRL 0x69 +#define DA9055_DAC_R_CTRL 0x6A +#define DA9055_HP_L_CTRL 0x6B +#define DA9055_HP_R_CTRL 0x6C +#define DA9055_LINE_CTRL 0x6D +#define DA9055_MIXOUT_L_CTRL 0x6E +#define DA9055_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA9055_LDO_CTRL 0x90 +#define DA9055_IO_CTRL 0x91 +#define DA9055_GAIN_RAMP_CTRL 0x92 +#define DA9055_MIC_CONFIG 0x93 +#define DA9055_PC_COUNT 0x94 +#define DA9055_CP_VOL_THRESHOLD1 0x95 +#define DA9055_CP_DELAY 0x96 +#define DA9055_CP_DETECTOR 0x97 +#define DA9055_AIF_OFFSET 0x98 +#define DA9055_DIG_CTRL 0x99 +#define DA9055_ALC_CTRL2 0x9A +#define DA9055_ALC_CTRL3 0x9B +#define DA9055_ALC_NOISE 0x9C +#define DA9055_ALC_TARGET_MIN 0x9D +#define DA9055_ALC_TARGET_MAX 0x9E +#define DA9055_ALC_GAIN_LIMITS 0x9F +#define DA9055_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA9055_ALC_ANTICLIP_CTRL 0xA1 +#define DA9055_ALC_ANTICLIP_LEVEL 0xA2 +#define DA9055_ALC_OFFSET_OP2M_L 0xA6 +#define DA9055_ALC_OFFSET_OP2U_L 0xA7 +#define DA9055_ALC_OFFSET_OP2M_R 0xAB +#define DA9055_ALC_OFFSET_OP2U_R 0xAC +#define DA9055_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA9055_ALC_CIC_OP_LVL_DATA 0xAE +#define DA9055_DAC_NG_SETUP_TIME 0xAF +#define DA9055_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA9055_DAC_NG_ON_THRESHOLD 0xB1 +#define DA9055_DAC_NG_CTRL 0xB2 + +/* DIG_ROUTING_AIF bit fields */ +#define DA9055_AIF_L_SRC (0 << 0) +#define DA9055_AIF_R_SRC (1 << 4) + +/* SR bit fields */ +#define DA9055_SR_8000 (0x1 << 0) +#define DA9055_SR_11025 (0x2 << 0) +#define DA9055_SR_12000 (0x3 << 0) +#define DA9055_SR_16000 (0x5 << 0) +#define DA9055_SR_22050 (0x6 << 0) +#define DA9055_SR_24000 (0x7 << 0) +#define DA9055_SR_32000 (0x9 << 0) +#define DA9055_SR_44100 (0xA << 0) +#define DA9055_SR_48000 (0xB << 0) +#define DA9055_SR_88200 (0xE << 0) +#define DA9055_SR_96000 (0xF << 0) + +/* REFERENCES bit fields */ +#define DA9055_BIAS_EN (1 << 3) +#define DA9055_VMID_EN (1 << 7) + +/* PLL_CTRL bit fields */ +#define DA9055_PLL_INDIV_10_20_MHZ (1 << 2) +#define DA9055_PLL_SRM_EN (1 << 6) +#define DA9055_PLL_EN (1 << 7) + +/* AIF_CLK_MODE bit fields */ +#define DA9055_AIF_BCLKS_PER_WCLK_32 (0 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_64 (1 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_128 (2 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_256 (3 << 0) +#define DA9055_AIF_CLK_EN_SLAVE_MODE (0 << 7) +#define DA9055_AIF_CLK_EN_MASTER_MODE (1 << 7) + +/* AIF_CTRL bit fields */ +#define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) +#define DA9055_AIF_FORMAT_LEFT_J (1 << 0) +#define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_WORD_S16_LE (0 << 2) +#define DA9055_AIF_WORD_S20_3LE (1 << 2) +#define DA9055_AIF_WORD_S24_LE (2 << 2) +#define DA9055_AIF_WORD_S32_LE (3 << 2) +#define DA9055_AIF_OE (1 << 6) +#define DA9055_AIF_EN (1 << 7) + +/* CP_CTRL bit fields */ +#define DA9055_CP_MCHANGE (3 << 4) +#define DA9055_CP_EN (1 << 7) + +/* MIC_BIAS_CTRL bit fields */ +#define DA9055_MIC_BIAS_EN (1 << 7) + +/* MIXIN_L_CTRL bit fields */ +#define DA9055_MIXIN_L_MIX_EN (1 << 3) + +/* MIXIN_R_CTRL bit fields */ +#define DA9055_MIXIN_R_MIX_EN (1 << 3) + +/* ADC_L_CTRL bit fields */ +#define DA9055_ADC_L_EN (1 << 7) + +/* ADC_R_CTRL bit fields */ +#define DA9055_ADC_R_EN (1 << 7) + +/* DAC_L_CTRL bit fields */ +#define DA9055_DAC_L_MUTE_EN (1 << 6) + +/* DAC_R_CTRL bit fields */ +#define DA9055_DAC_R_MUTE_EN (1 << 6) + +/* HP_L_CTRL bit fields */ +#define DA9055_HP_L_AMP_OE (1 << 3) + +/* HP_R_CTRL bit fields */ +#define DA9055_HP_R_AMP_OE (1 << 3) + +/* LINE_CTRL bit fields */ +#define DA9055_LINE_AMP_OE (1 << 3) + +/* MIXOUT_L_CTRL bit fields */ +#define DA9055_MIXOUT_L_MIX_EN (1 << 3) +#define DA9055_MIXOUT_L_AMP_EN (1 << 7) + +/* MIXOUT_R_CTRL bit fields */ +#define DA9055_MIXOUT_R_MIX_EN (1 << 3) +#define DA9055_MIXOUT_R_AMP_EN (1 << 7) + +/* ALC_CIC_OP_LEVEL_CTRL bit fields */ +#define DA9055_ALC_CAPTURE_DATA (0 << 0) +#define DA9055_ALC_DATA_BOT (1 << 0) +#define DA9055_ALC_DATA_MIDDLE (2 << 0) +#define DA9055_ALC_DATA_TOP (3 << 0) +#define DA9055_ALC_CIC_OP_CHANNEL_LEFT (0 << 7) +#define DA9055_ALC_CIC_OP_CHANNEL_RIGHT (1 << 7) + +#define DA9055_GAIN_RAMPING_EN (1 << 5) + +#define DA9055_ALC_OFFSET_15_8 0x00FF00 +#define DA9055_ALC_OFFSET_17_16 0x030000 +#define DA9055_ALC_AVG_ITERATIONS 5 + +struct pll_div { + int fref; + int fout; + u8 frac_top; + u8 frac_bot; + u8 integer; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL divisor table */ +static const struct pll_div da9055_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz and its harmonics */ + {11289600, 2822400, 0x00, 0x00, 0x20, 1}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x03, 0x61, 0x1E, 1}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x0C, 0xCC, 0x1D, 1}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x19, 0x45, 0x1B, 1}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x18, 0x56, 0x1A, 1}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x02, 0xD0, 0x19, 1}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x1A, 0x1C, 0x12, 1}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x0B, 0x6D, 0x12, 1}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x07, 0xDD, 0x12, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz and its harmonics */ + {11289600, 3072000, 0x1A, 0x8E, 0x22, 1}, /* MCLK=11.2896Mhz */ + {12000000, 3072000, 0x18, 0x93, 0x20, 1}, /* MCLK=12Mhz */ + {12288000, 3072000, 0x00, 0x00, 0x20, 1}, /* MCLK=12.288Mhz */ + {13000000, 3072000, 0x07, 0xEA, 0x1E, 1}, /* MCLK=13Mhz */ + {13500000, 3072000, 0x04, 0x11, 0x1D, 1}, /* MCLK=13.5Mhz */ + {14400000, 3072000, 0x09, 0xD0, 0x1B, 1}, /* MCLK=14.4Mhz */ + {19200000, 3072000, 0x0F, 0x5C, 0x14, 1}, /* MCLK=19.2Mhz */ + {19680000, 3072000, 0x1F, 0x60, 0x13, 1}, /* MCLK=19.68Mhz */ + {19800000, 3072000, 0x1B, 0x80, 0x13, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + {11289600, 2822400, 0x0D, 0x47, 0x21, 0}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x0D, 0xFA, 0x1F, 0}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x16, 0x66, 0x1E, 0}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x00, 0x98, 0x1D, 0}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x1E, 0x33, 0x1B, 0}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x06, 0x50, 0x1A, 0}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x14, 0xBC, 0x13, 0}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x05, 0x66, 0x13, 0}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x01, 0xAE, 0x13, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA9055_CLKSRC_MCLK +}; + +/* Gain and Volume */ + +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da9055_hpf_cutoff_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da9055_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +static const struct soc_enum da9055_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da9055_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da9055_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +static const struct soc_enum da9055_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +/* Gain ramping rate value */ +static const char * const da9055_gain_ramping_txt[] = { + "nominal rate", "nominal rate * 4", "nominal rate * 8", + "nominal rate / 8" +}; + +static const struct soc_enum da9055_gain_ramping_rate = + SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); + +/* DAC noise gate setup time value */ +static const char * const da9055_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da9055_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, + da9055_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da9055_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, + da9055_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da9055_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, + da9055_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da9055_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da9055_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, + da9055_dac_soft_mute_rate_txt); + +/* LDO voltage level value */ +static const char * const da9055_ldo_lvl_select_txt[] = { + "1.05V", "1.1V", "1.2V", "1.4V" +}; + +static const struct soc_enum da9055_ldo_lvl_select = + SOC_ENUM_SINGLE(DA9055_LDO_CTRL, 4, 4, da9055_ldo_lvl_select_txt); + +/* AIF routing select */ +static const char * const da9055_aif_select_txt[] = { + "ADC left", "ADC right", "AIF input left", "AIF input right" +}; + +static const struct soc_enum da9055_aif_l_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_AIF, 0, 4, da9055_aif_select_txt); + +static const struct soc_enum da9055_aif_r_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_AIF, 4, 4, da9055_aif_select_txt); + +/* DAC routing select */ +static const char * const da9055_dac_select_txt[] = { + "ADC left output", "ADC right output", "AIF input left", + "AIF input right" +}; + +static const struct soc_enum da9055_dac_l_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_select_txt); + +static const struct soc_enum da9055_dac_r_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_select_txt); + +/* Digital MIC clock rate select */ +static const char * const da9055_dmic_clk_rate_txt[] = { + "3MHz", "1.5MHz" +}; + +static const struct soc_enum da9055_dmic_clk_rate = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 2, 2, da9055_dmic_clk_rate_txt); + +/* Digital MIC sample phase select */ +static const char * const da9055_dmic_phase_txt[] = { + "Sample on DMICCLK edges", "Sample between DMICCLK edges" +}; + +static const struct soc_enum da9055_dmic_phase = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 1, 2, da9055_dmic_phase_txt); + +/* Digital MIC channel select */ +static const char * const da9055_dmic_channel_select_txt[] = { + "Rising Left Falling Right", "Falling Left Rising Right" +}; + +static const struct soc_enum da9055_dmic_channel_select = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 0, 2, + da9055_dmic_channel_select_txt); + +/* MIC bias select */ +static const char * const da9055_mic_bias_select_txt[] = { + "MIC Bias 1", "MIC Bias 2" +}; + +static const struct soc_enum da9055_mic_bias_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 6, 2, + da9055_mic_bias_select_txt); + +/* MIC bias voltage level select */ +static const char * const da9055_mic_bias_level_txt[] = { + "1.6V", "1.8V", "2.1V", "2.2V" +}; + +static const struct soc_enum da9055_mic_bias_level = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 4, 4, da9055_mic_bias_level_txt); + +/* MIC Left Amp routing select */ +static const char * const da9055_mic_l_select_txt[] = { + "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_l_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_select_txt); + +/* MIC Right Amp routing select */ +static const char * const da9055_mic_r_select_txt[] = { + "MIC2_R_L", "MIC2_R", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_r_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_select_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da9055_signal_tracking_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da9055_integ_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, + da9055_signal_tracking_rate_txt); + +static const struct soc_enum da9055_integ_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, + da9055_signal_tracking_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da9055_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da9055_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da9055_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da9055_hold_time = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); + +static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA9055_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA9055_ALC_AVG_ITERATIONS; +} + +static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 reg_val, adc_left, adc_right; + int avg_left_data, avg_right_data, offset_l, offset_r; + + if (ucontrol->value.integer.value[0]) { + /* + * While enabling ALC (or ALC sync mode), calibration of the DC + * offsets must be done first + */ + + /* Save current values from ADC control registers */ + adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); + adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_ADC_L_EN, DA9055_ADC_L_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_ADC_R_EN, DA9055_ADC_R_EN); + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_L, reg_val); + reg_val = (offset_l & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_L, reg_val); + + reg_val = (offset_r & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_R, reg_val); + reg_val = (offset_r & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_R, reg_val); + + /* Restore original values of ADC control registers */ + snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); + snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +static const struct snd_kcontrol_new da9055_snd_controls[] = { + + /* Volume controls */ + SOC_DOUBLE_R_TLV("Mic Volume", + DA9055_MIC_L_GAIN, DA9055_MIC_R_GAIN, + 0, 0x7, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", + DA9055_AUX_L_GAIN, DA9055_AUX_R_GAIN, + 0, 0x3f, 0, aux_vol_tlv), + SOC_DOUBLE_R_TLV("Mixin PGA Volume", + DA9055_MIXIN_L_GAIN, DA9055_MIXIN_R_GAIN, + 0, 0xf, 0, mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", + DA9055_ADC_L_GAIN, DA9055_ADC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Volume", + DA9055_DAC_L_GAIN, DA9055_DAC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + SOC_DOUBLE_R_TLV("HeadPhone Volume", + DA9055_HP_L_GAIN, DA9055_HP_R_GAIN, + 0, 0x3f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA9055_LINE_GAIN, 0, 0x3f, 0, + lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA9055_DAC_FILTERS4, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA9055_DAC_FILTERS2, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA9055_DAC_FILTERS2, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA9055_DAC_FILTERS3, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA9055_DAC_FILTERS3, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA9055_DAC_FILTERS4, 0, 0xf, 0, + eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA9055_ADC_FILTERS1, 7, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da9055_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA9055_ADC_FILTERS1, 3, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da9055_adc_vf_cutoff), + + SOC_SINGLE("DAC HPF Switch", DA9055_DAC_FILTERS1, 7, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da9055_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA9055_DAC_FILTERS1, 3, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da9055_dac_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Mute Switch", DA9055_MIC_L_CTRL, + DA9055_MIC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Aux Mute Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Mixin PGA Mute Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("ADC Mute Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("HeadPhone Mute Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 6, 1, 0), + SOC_SINGLE("Lineout Mute Switch", DA9055_LINE_CTRL, 6, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA9055_DAC_FILTERS5, 7, 1, 0), + SOC_ENUM("DAC Soft Mute Rate", da9055_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("HeadPhone ZC Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 4, 1, 0), + SOC_SINGLE("Lineout ZC Switch", DA9055_LINE_CTRL, 4, 1, 0), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA9055_DAC_L_CTRL, + DA9055_DAC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("HeadPhone Gain Ramping Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 5, 1, 0), + SOC_SINGLE("Lineout Gain Ramping Switch", DA9055_LINE_CTRL, 5, 1, 0), + SOC_ENUM("Gain Ramping Rate", da9055_gain_ramping_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Enable Switch", DA9055_DAC_NG_CTRL, 7, 1, 0), + SOC_SINGLE("DAC NG ON Threshold", DA9055_DAC_NG_ON_THRESHOLD, + 0, 0x7, 0), + SOC_SINGLE("DAC NG OFF Threshold", DA9055_DAC_NG_OFF_THRESHOLD, + 0, 0x7, 0), + SOC_ENUM("DAC NG Setup Time", da9055_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da9055_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da9055_dac_ng_rampdown_rate), + + /* LDO control */ + SOC_SINGLE("LDO Enable", DA9055_LDO_CTRL, 7, 1, 0), + SOC_ENUM("LDO Level Select", da9055_ldo_lvl_select), + + /* AIF Routing control */ + SOC_ENUM("AIF Left Select", da9055_aif_l_select), + SOC_ENUM("AIF Right Select", da9055_aif_r_select), + + /* DAC Routing control */ + SOC_ENUM("DAC Left Select", da9055_dac_l_select), + SOC_ENUM("DAC Right Select", da9055_dac_r_select), + + /* DAC Invertion control */ + SOC_SINGLE("DAC Left Invert", DA9055_DIG_CTRL, 3, 1, 0), + SOC_SINGLE("DAC Right Invert", DA9055_DIG_CTRL, 7, 1, 0), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Enable", DA9055_MIXIN_L_SELECT, + DA9055_MIXIN_R_SELECT, 7, 1, 0), + SOC_ENUM("DMIC Clock Rate", da9055_dmic_clk_rate), + SOC_ENUM("DMIC Sample Phase", da9055_dmic_phase), + SOC_ENUM("DMIC Channel Select", da9055_dmic_channel_select), + + /* MIC PGA input selection controls */ + SOC_ENUM("Mic Left Input Select", da9055_mic_l_select), + SOC_ENUM("Mic Right Input Select", da9055_mic_r_select), + + /* To select if MIC Bias powers MIC 1 or MIC 2 */ + SOC_ENUM("Mic Bias Select", da9055_mic_bias_select), + + /* MIC Bias voltage select control */ + SOC_ENUM("Mic Bias Level", da9055_mic_bias_level), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Enable", DA9055_ALC_CTRL1, 3, 7, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE_EXT("ALC Sync Mode Enable", DA9055_ALC_CTRL1, 1, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE("ALC Offset Enable", DA9055_ALC_CTRL1, 0, 1, 0), + SOC_SINGLE("ALC Anticlip Mode Enable", DA9055_ALC_ANTICLIP_CTRL, + 7, 1, 0), + SOC_SINGLE("ALC Anticlip Level", DA9055_ALC_ANTICLIP_LEVEL, + 0, 0x7f, 0), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA9055_ALC_TARGET_MIN, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA9055_ALC_TARGET_MAX, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA9055_ALC_NOISE, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA9055_ALC_GAIN_LIMITS, + 4, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA9055_ALC_GAIN_LIMITS, + 0, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 0, 0x7, 0, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 4, 0x7, 0, alc_analog_gain_tlv), + SOC_ENUM("ALC Attack Rate", da9055_attack_rate), + SOC_ENUM("ALC Release Rate", da9055_release_rate), + SOC_ENUM("ALC Hold Time", da9055_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da9055_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da9055_integ_release_rate), +}; + +/* DAPM Controls */ + +/* In Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXIN_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_L_SELECT, 2, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXIN_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXIN_R_SELECT, 3, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXOUT_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_L_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA9055_MIXOUT_L_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_L_SELECT, + 6, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXOUT_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA9055_MIXOUT_R_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_R_SELECT, + 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { + /* Input Side */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2L"), + SND_SOC_DAPM_INPUT("MIC2R"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA9055_MIC_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA9055_MIC_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left", DA9055_AUX_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right", DA9055_AUX_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Left", DA9055_MIXIN_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Right", DA9055_MIXIN_R_CTRL, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinl_controls[0], + ARRAY_SIZE(da9055_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinr_controls[0], + ARRAY_SIZE(da9055_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA9055_ADC_L_CTRL, 7, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA9055_ADC_R_CTRL, 7, 0), + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA9055_DAC_L_CTRL, 7, 0), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA9055_DAC_R_CTRL, 7, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutl_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutr_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Lineout", DA9055_LINE_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA9055_HP_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA9055_HP_R_CTRL, 7, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da9055_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"Mic Left", NULL, "MIC1"}, + {"Mic Left", NULL, "MIC2L"}, + {"Mic Right", NULL, "MIC2L"}, + {"Mic Right", NULL, "MIC2R"}, + {"Aux Left", NULL, "AUXL"}, + {"Aux Right", NULL, "AUXR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux Left Switch", "Aux Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux Right Switch", "Aux Right"}, + {"In Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + + {"MIXIN Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "MIXIN Left"}, + + {"MIXIN Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "MIXIN Right"}, + + /* Output path */ + {"Out Mixer Left", "Aux Left Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Left", "Aux Left Invert Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux Right Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Right", "Aux Right Invert Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Headphone Left", NULL, "Out Mixer Left"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "Out Mixer Right"}, + {"HPR", NULL, "Headphone Right"}, + + {"Lineout", NULL, "Out Mixer Right"}, + {"LINE", NULL, "Lineout"}, +}; + +/* Codec private data */ +struct da9055_priv { + struct regmap *regmap; + unsigned int mclk_rate; + int master; +}; + +static struct reg_default da9055_reg_defaults[] = { + { 0x21, 0x10 }, + { 0x22, 0x0A }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x0C }, + { 0x28, 0x01 }, + { 0x29, 0x08 }, + { 0x2A, 0x32 }, + { 0x2B, 0x00 }, + { 0x30, 0x35 }, + { 0x31, 0x35 }, + { 0x32, 0x00 }, + { 0x33, 0x00 }, + { 0x34, 0x03 }, + { 0x35, 0x03 }, + { 0x36, 0x6F }, + { 0x37, 0x6F }, + { 0x38, 0x80 }, + { 0x39, 0x01 }, + { 0x3A, 0x01 }, + { 0x40, 0x00 }, + { 0x41, 0x88 }, + { 0x42, 0x88 }, + { 0x43, 0x08 }, + { 0x44, 0x80 }, + { 0x45, 0x6F }, + { 0x46, 0x6F }, + { 0x47, 0x61 }, + { 0x48, 0x35 }, + { 0x49, 0x35 }, + { 0x4A, 0x35 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x60, 0x44 }, + { 0x61, 0x44 }, + { 0x62, 0x00 }, + { 0x63, 0x40 }, + { 0x64, 0x40 }, + { 0x65, 0x40 }, + { 0x66, 0x40 }, + { 0x67, 0x40 }, + { 0x68, 0x40 }, + { 0x69, 0x48 }, + { 0x6A, 0x40 }, + { 0x6B, 0x41 }, + { 0x6C, 0x40 }, + { 0x6D, 0x40 }, + { 0x6E, 0x10 }, + { 0x6F, 0x10 }, + { 0x90, 0x80 }, + { 0x92, 0x02 }, + { 0x93, 0x00 }, + { 0x99, 0x00 }, + { 0x9A, 0x00 }, + { 0x9B, 0x00 }, + { 0x9C, 0x3F }, + { 0x9D, 0x00 }, + { 0x9E, 0x3F }, + { 0x9F, 0xFF }, + { 0xA0, 0x71 }, + { 0xA1, 0x00 }, + { 0xA2, 0x00 }, + { 0xA6, 0x00 }, + { 0xA7, 0x00 }, + { 0xAB, 0x00 }, + { 0xAC, 0x00 }, + { 0xAD, 0x00 }, + { 0xAF, 0x08 }, + { 0xB0, 0x00 }, + { 0xB1, 0x00 }, + { 0xB2, 0x00 }, +}; + +static bool da9055_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case DA9055_STATUS1: + case DA9055_PLL_STATUS: + case DA9055_AUX_L_GAIN_STATUS: + case DA9055_AUX_R_GAIN_STATUS: + case DA9055_MIC_L_GAIN_STATUS: + case DA9055_MIC_R_GAIN_STATUS: + case DA9055_MIXIN_L_GAIN_STATUS: + case DA9055_MIXIN_R_GAIN_STATUS: + case DA9055_ADC_L_GAIN_STATUS: + case DA9055_ADC_R_GAIN_STATUS: + case DA9055_DAC_L_GAIN_STATUS: + case DA9055_DAC_R_GAIN_STATUS: + case DA9055_HP_L_GAIN_STATUS: + case DA9055_HP_R_GAIN_STATUS: + case DA9055_LINE_GAIN_STATUS: + case DA9055_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +/* Set DAI word length */ +static int da9055_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_ctrl, fs; + u32 sysclk; + + /* Set AIF source to Left and Right ADC */ + snd_soc_write(codec, DA9055_DIG_ROUTING_AIF, + DA9055_AIF_L_SRC | DA9055_AIF_R_SRC); + + aif_ctrl = snd_soc_read(codec, DA9055_AIF_CTRL) & 0xf3; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif_ctrl |= DA9055_AIF_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif_ctrl |= DA9055_AIF_WORD_S20_3LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif_ctrl |= DA9055_AIF_WORD_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif_ctrl |= DA9055_AIF_WORD_S32_LE; + break; + default: + return -EINVAL; + } + + aif_ctrl |= (DA9055_AIF_OE | DA9055_AIF_EN); + + /* Enable AIF */ + snd_soc_write(codec, DA9055_AIF_CTRL, aif_ctrl); + + switch (params_rate(params)) { + case 8000: + fs = DA9055_SR_8000; + sysclk = 3072000; + break; + case 11025: + fs = DA9055_SR_11025; + sysclk = 2822400; + break; + case 12000: + fs = DA9055_SR_12000; + sysclk = 3072000; + break; + case 16000: + fs = DA9055_SR_16000; + sysclk = 3072000; + break; + case 22050: + fs = DA9055_SR_22050; + sysclk = 2822400; + break; + case 32000: + fs = DA9055_SR_32000; + sysclk = 3072000; + break; + case 44100: + fs = DA9055_SR_44100; + sysclk = 2822400; + break; + case 48000: + fs = DA9055_SR_48000; + sysclk = 3072000; + break; + case 88200: + fs = DA9055_SR_88200; + sysclk = 2822400; + break; + case 96000: + fs = DA9055_SR_96000; + sysclk = 3072000; + break; + default: + return -EINVAL; + } + + if (da9055->mclk_rate) { + /* PLL Mode, Write actual FS */ + snd_soc_write(codec, DA9055_SR, fs); + } else { + /* + * Non-PLL Mode + * When PLL is bypassed, chip assumes constant MCLK of + * 12.288MHz and uses sample rate value to divide this MCLK + * to derive its sys clk. As sys clk has to be 256 * Fs, we + * need to write constant sample rate i.e. 48KHz. + */ + snd_soc_write(codec, DA9055_SR, DA9055_SR_48000); + } + + if (da9055->mclk_rate && (da9055->mclk_rate != sysclk)) { + /* PLL Mode */ + if (!da9055->master) { + /* PLL slave mode, enable PLL and also SRM */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN | DA9055_PLL_SRM_EN, + DA9055_PLL_EN | DA9055_PLL_SRM_EN); + } else { + /* PLL master mode, only enable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN, DA9055_PLL_EN); + } + } else { + /* Non PLL Mode, disable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + } + + return 0; +} + +/* Set DAI mode and Format */ +static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_clk_mode, aif_ctrl; + + aif_clk_mode = snd_soc_read(codec, DA9055_AIF_CLK_MODE) & 0x7c; + aif_ctrl = snd_soc_read(codec, DA9055_AIF_CTRL) & 0xfc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* DA9055 in I2S Master Mode */ + da9055->master = 1; + aif_clk_mode |= DA9055_AIF_CLK_EN_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* DA9055 in I2S Slave Mode */ + da9055->master = 0; + aif_clk_mode |= DA9055_AIF_CLK_EN_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_ctrl |= DA9055_AIF_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_ctrl |= DA9055_AIF_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif_ctrl |= DA9055_AIF_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + aif_clk_mode |= DA9055_AIF_BCLKS_PER_WCLK_32; + + snd_soc_write(codec, DA9055_AIF_CLK_MODE, aif_clk_mode); + snd_soc_write(codec, DA9055_AIF_CTRL, aif_ctrl); + + return 0; +} + +static int da9055_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, DA9055_DAC_R_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, 0); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, 0); + } + + return 0; +} + +#define DA9055_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da9055_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA9055_CLKSRC_MCLK: + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da9055->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* + * da9055_set_dai_pll : Configure the codec PLL + * @param codec_dai : Pointer to codec DAI + * @param pll_id : da9055 has only one pll, so pll_id is always zero + * @param fref : Input MCLK frequency + * @param fout : FsDM value + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 11.2896MHz, 12MHz, 12.288MHz, + * 13MHz, 13.5MHz, 14.4MHz, 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da9055_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + u8 pll_frac_top, pll_frac_bot, pll_integer, cnt; + + /* Disable PLL before setting the divisors */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + + /* In slave mode, there is only one set of divisors */ + if (!da9055->master) + fout = 2822400; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da9055_pll_div); cnt++) { + /* Check fref, mode and fout */ + if ((fref == da9055_pll_div[cnt].fref) && + (da9055->master == da9055_pll_div[cnt].mode) && + (fout == da9055_pll_div[cnt].fout)) { + /* All match, pick up divisors */ + pll_frac_top = da9055_pll_div[cnt].frac_top; + pll_frac_bot = da9055_pll_div[cnt].frac_bot; + pll_integer = da9055_pll_div[cnt].integer; + break; + } + } + if (cnt >= ARRAY_SIZE(da9055_pll_div)) + goto pll_err; + + /* Write PLL dividers */ + snd_soc_write(codec, DA9055_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA9055_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA9055_PLL_INTEGER, pll_integer); + + return 0; +pll_err: + dev_err(codec_dai->dev, "Error in setting up PLL\n"); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da9055_dai_ops = { + .hw_params = da9055_hw_params, + .set_fmt = da9055_set_dai_fmt, + .set_sysclk = da9055_set_dai_sysclk, + .set_pll = da9055_set_dai_pll, + .digital_mute = da9055_mute, +}; + +static struct snd_soc_dai_driver da9055_dai = { + .name = "da9055-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + .ops = &da9055_dai_ops, + .symmetric_rates = 1, +}; + +static int da9055_probe(struct snd_soc_codec *codec) +{ + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = da9055->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + da9055->mclk_rate = 0; /* This will be set from set_sysclk() */ + da9055->master = 0; /* This will be set from set_fmt() */ + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_AUX_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + + /* Input side settings */ + + /* Enable VMID reference & master bias */ + snd_soc_write(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN); + + /* Enable Mic Bias */ + snd_soc_write(codec, DA9055_MIC_BIAS_CTRL, DA9055_MIC_BIAS_EN); + + /* Enable Left and Right input mixers */ + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_L_MIX_EN, DA9055_MIXIN_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_MIXIN_R_MIX_EN, DA9055_MIXIN_R_MIX_EN); + + /* Output side settings */ + + /* Enable Left and Right output mixers */ + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_MIX_EN, DA9055_MIXOUT_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_MIX_EN, DA9055_MIXOUT_R_MIX_EN); + + /* Enable Left and Right output Amps */ + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_AMP_EN, DA9055_MIXOUT_L_AMP_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_AMP_EN, DA9055_MIXOUT_R_AMP_EN); + + /* Enable the charge pump and set mode */ + snd_soc_update_bits(codec, DA9055_CP_CTRL, + DA9055_CP_EN | DA9055_CP_MCHANGE, + DA9055_CP_EN | DA9055_CP_MCHANGE); + + /* Enable Left and Right HeadPhone Output */ + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_HP_L_AMP_OE, DA9055_HP_L_AMP_OE); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_HP_R_AMP_OE, DA9055_HP_R_AMP_OE); + + /* Enable Lineout Output Enable */ + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_LINE_AMP_OE, DA9055_LINE_AMP_OE); + + /* Set this as per your system configuration */ + snd_soc_write(codec, DA9055_PLL_CTRL, DA9055_PLL_INDIV_10_20_MHZ); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da9055 = { + .probe = da9055_probe, + + .controls = da9055_snd_controls, + .num_controls = ARRAY_SIZE(da9055_snd_controls), + + .dapm_widgets = da9055_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da9055_dapm_widgets), + .dapm_routes = da9055_audio_map, + .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), +}; + +static const struct regmap_config da9055_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da9055_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da9055_reg_defaults), + .volatile_reg = da9055_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da9055_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da9055_priv *da9055; + int ret; + + da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + GFP_KERNEL); + if (!da9055) + return -ENOMEM; + + i2c_set_clientdata(i2c, da9055); + + da9055->regmap = regmap_init_i2c(i2c, &da9055_regmap_config); + if (IS_ERR(da9055->regmap)) { + ret = PTR_ERR(da9055->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da9055, &da9055_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + ret); + regmap_exit(da9055->regmap); + } + return ret; +} + +static int __devexit da9055_remove(struct i2c_client *client) +{ + struct da9055_priv *da9055 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(da9055->regmap); + return 0; +} + +static const struct i2c_device_id da9055_i2c_id[] = { + { "da9055", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da9055_i2c_driver = { + .driver = { + .name = "da9055", + .owner = THIS_MODULE, + }, + .probe = da9055_i2c_probe, + .remove = __devexit_p(da9055_remove), + .id_table = da9055_i2c_id, +}; + +module_i2c_driver(da9055_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); +MODULE_AUTHOR("David Chen, Ashish Chavan"); +MODULE_LICENSE("GPL"); -- 1.7.1

3 6

[alsa-devel] [PATCH] ASoC: ad1836: Convert to direct regmap usage.
by Lars-Peter Clausen 13 Sep '12

13 Sep '12

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 76 +++++++++++++++++++++++++++++++-------------- 1 file changed, 53 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ae1eb51..dce6ebe 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -19,6 +19,8 @@ #include <sound/soc.h> #include <sound/tlv.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> + #include "ad1836.h" enum ad1836_type { @@ -30,6 +32,7 @@ enum ad1836_type { /* codec private data */ struct ad1836_priv { enum ad1836_type type; + struct regmap *regmap; }; /* @@ -161,8 +164,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(dai->codec); int word_len = 0; - struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { @@ -178,10 +181,12 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1, + AD1836_DAC_WORD_LEN_MASK, word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, + AD1836_ADC_WORD_LEN_MASK, word_len << AD1836_ADC_WORD_OFFSET); return 0; @@ -223,15 +228,17 @@ static struct snd_soc_dai_driver ad183x_dais[] = { #ifdef CONFIG_PM static int ad1836_suspend(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* reset clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, 0); } static int ad1836_resume(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* restore clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX); } #else @@ -250,37 +257,30 @@ static int ad1836_probe(struct snd_soc_codec *codec) num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2; num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2; - ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); - return ret; - } - /* default setting for ad1836 */ /* de-emphasis: 48kHz, power-on dac */ - snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); + regmap_write(ad1836->regmap, AD1836_DAC_CTRL1, 0x300); /* unmute dac channels */ - snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); + regmap_write(ad1836->regmap, AD1836_DAC_CTRL2, 0x0); /* high-pass filter enable, power-on adc */ - snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL1, 0x100); /* unmute adc channles, adc aux mode */ - snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL2, 0x180); /* volume */ for (i = 1; i <= num_dacs; ++i) { - snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF); - snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF); + regmap_write(ad1836->regmap, AD1836_DAC_L_VOL(i), 0x3FF); + regmap_write(ad1836->regmap, AD1836_DAC_R_VOL(i), 0x3FF); } if (ad1836->type == AD1836) { /* left/right diff:PGA/MUX */ - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL3, 0x3A); ret = snd_soc_add_codec_controls(codec, ad1836_controls, ARRAY_SIZE(ad1836_controls)); if (ret) return ret; } else { - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL3, 0x00); } ret = snd_soc_add_codec_controls(codec, ad183x_dac_controls, num_dacs * 2); @@ -313,8 +313,9 @@ static int ad1836_probe(struct snd_soc_codec *codec) /* power down chip */ static int ad1836_remove(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* reset clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, 0); } @@ -323,8 +324,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { .remove = ad1836_remove, .suspend = ad1836_suspend, .resume = ad1836_resume, - .reg_cache_size = AD1836_NUM_REGS, - .reg_word_size = sizeof(u16), .controls = ad183x_controls, .num_controls = ARRAY_SIZE(ad183x_controls), @@ -334,6 +333,33 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { .num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes), }; +static const struct reg_default ad1836_reg_defaults[] = { + { AD1836_DAC_CTRL1, 0x0000 }, + { AD1836_DAC_CTRL2, 0x0000 }, + { AD1836_DAC_L_VOL(0), 0x0000 }, + { AD1836_DAC_R_VOL(0), 0x0000 }, + { AD1836_DAC_L_VOL(1), 0x0000 }, + { AD1836_DAC_R_VOL(1), 0x0000 }, + { AD1836_DAC_L_VOL(2), 0x0000 }, + { AD1836_DAC_R_VOL(2), 0x0000 }, + { AD1836_DAC_L_VOL(3), 0x0000 }, + { AD1836_DAC_R_VOL(3), 0x0000 }, + { AD1836_ADC_CTRL1, 0x0000 }, + { AD1836_ADC_CTRL2, 0x0000 }, + { AD1836_ADC_CTRL3, 0x0000 }, +}; + +static const struct regmap_config ad1836_regmap_config = { + .val_bits = 12, + .reg_bits = 4, + .read_flag_mask = 0x08, + + .max_register = AD1836_ADC_CTRL3, + .reg_defaults = ad1836_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ad1836_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct ad1836_priv *ad1836; @@ -344,6 +370,10 @@ static int __devinit ad1836_spi_probe(struct spi_device *spi) if (ad1836 == NULL) return -ENOMEM; + ad1836->regmap = devm_regmap_init_spi(spi, &ad1836_regmap_config); + if (IS_ERR(ad1836->regmap)) + return PTR_ERR(ad1836->regmap); + ad1836->type = spi_get_device_id(spi)->driver_data; spi_set_drvdata(spi, ad1836); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: ad193x: Use managed regmap init
by Lars-Peter Clausen 13 Sep '12

13 Sep '12

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 50 +++++++++------------------------------------ 1 file changed, 10 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 13e62be..2f75266 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -381,40 +381,25 @@ static const struct regmap_config ad193x_spi_regmap_config = { static int __devinit ad193x_spi_probe(struct spi_device *spi) { struct ad193x_priv *ad193x; - int ret; ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config); - if (IS_ERR(ad193x->regmap)) { - ret = PTR_ERR(ad193x->regmap); - goto err_out; - } + ad193x->regmap = devm_regmap_init_spi(spi, &ad193x_spi_regmap_config); + if (IS_ERR(ad193x->regmap)) + return PTR_ERR(ad193x->regmap); spi_set_drvdata(spi, ad193x); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ad193x, &ad193x_dai, 1); - if (ret < 0) - goto err_regmap_exit; - - return 0; - -err_regmap_exit: - regmap_exit(ad193x->regmap); -err_out: - return ret; + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } static int __devexit ad193x_spi_remove(struct spi_device *spi) { - struct ad193x_priv *ad193x = spi_get_drvdata(spi); - snd_soc_unregister_codec(&spi->dev); - regmap_exit(ad193x->regmap); return 0; } @@ -449,40 +434,25 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct ad193x_priv *ad193x; - int ret; ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config); - if (IS_ERR(ad193x->regmap)) { - ret = PTR_ERR(ad193x->regmap); - goto err_out; - } + ad193x->regmap = devm_regmap_init_i2c(client, &ad193x_i2c_regmap_config); + if (IS_ERR(ad193x->regmap)) + return PTR_ERR(ad193x->regmap); i2c_set_clientdata(client, ad193x); - ret = snd_soc_register_codec(&client->dev, - &soc_codec_dev_ad193x, &ad193x_dai, 1); - if (ret < 0) - goto err_regmap_exit; - - return 0; - -err_regmap_exit: - regmap_exit(ad193x->regmap); -err_out: - return ret; + return snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } static int __devexit ad193x_i2c_remove(struct i2c_client *client) { - struct ad193x_priv *ad193x = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(ad193x->regmap); return 0; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH 00/11] ASoC: OMAP: Convert to use dmaengine
by Peter Ujfalusi 13 Sep '12

13 Sep '12

Hello, This series will switch the OMAP audio to use dmaengine. The final patch which does the switch was based on Russell King's earlier patch. The first 10 patch is to prepare the OMAP audio drivers for a smooth change to dmaengine: - sDMA FRAME sync mode is removed and replaced with PACKET mode - dai drivers no longer need to configure sDMA sync mode - dai drivers does not need to specify the DMA word length - with the exception of the omap-hdmi driver which requires 32bit word length regardless of the audio format in use - the McPDM driver used (to my surprise) hackish way of getting the DMA channel and address - via defines from some header files I have tested the series on: - BeagleBoard (audio via McBSP) with aplay/arecord. In element mode and in threshold mode with different period sizes - OMAP4 Blaze (audio via McPDM and DMIC) With this conversion the NO_PERIOD_WAKEUP is not supported at the moment. I'll be looking at the dmaengine core and omap parts to add this feature back. The patches has been generated against: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 Janusz: Can you retest this series on OMAP1 to be sure I have not broken it? Ricardo: Can you test the omap-hmdi if it is still working? Regards, Peter --- Peter Ujfalusi (11): dmaengine: omap: Support for element mode in cyclic DMA ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode ASoC: omap-pcm: Select sDMA synchronization based on packet_size ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges ASoC: omap-mcpdm: Use platform_get_resource_* to get resources ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type ASoC: omap-pcm, omap-hdmi: Change the use of omap_pcm_dma_data->data_type ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time ASoC: omap-pcm: Convert to use dmaengine arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + drivers/dma/omap-dma.c | 5 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 17 +- sound/soc/omap/omap-mcbsp.c | 60 +++---- sound/soc/omap/omap-mcpdm.c | 41 +++-- sound/soc/omap/omap-pcm.c | 246 ++++++++--------------------- sound/soc/omap/omap-pcm.h | 4 +- 9 files changed, 136 insertions(+), 251 deletions(-) -- 1.7.12

4 17

[alsa-devel] Invitation to connect on LinkedIn
by Vitaly Chernooky via LinkedIn 13 Sep '12

13 Sep '12

LinkedIn ------------ Vitaly Chernooky requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Vitaly Accept invitation from Vitaly Chernooky http://www.linkedin.com/e/jpxkk3-h71n762j-6i/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Vitaly Chernooky http://www.linkedin.com/e/jpxkk3-h71n762j-6i/rso/18329483/HTJa/name/1380363… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-h71n762j-6i/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [sound-unstable:master 118/143] sound/pci/hda/patch_cirrus.c:1414:10: sparse: Initializer entry defined twice
by Fengguang Wu 13 Sep '12

13 Sep '12

Hi Takashi, FYI, there are new sparse warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 82cb521360dfd32f4f03ceaa74ef6b83ae983593 commit: ef596a57b4d7d8b258beb570ed309ef85bf24dd1 [118/143] ALSA: hda - Add support for MacBook Pro 10,1 All sparse warnings: + sound/pci/hda/patch_cirrus.c:1414:10: sparse: Initializer entry defined twice sound/pci/hda/patch_cirrus.c:1418:10: also defined here The root cause is, CS420X_MBP101 == CS420X_GPIO_23: @@ -85,6 +85,8 @@ enum { => CS420X_GPIO_23, CS420X_IMAC27_122 = CS420X_GPIO_23, CS420X_APPLE = CS420X_GPIO_13, => + CS420X_MBP101, + CS420X_MBP101_COEF, CS420X_AUTO, }; vim +1414 sound/pci/hda/patch_cirrus.c 1404 [CS420X_IMAC27] = { 1405 .type = HDA_FIXUP_PINS, 1406 .v.pins = imac27_pincfgs, 1407 .chained = true, 1408 .chain_id = CS420X_GPIO_13, 1409 }, 1410 [CS420X_GPIO_13] = { 1411 .type = HDA_FIXUP_FUNC, 1412 .v.func = cs420x_fixup_gpio_13, 1413 }, > 1414 [CS420X_GPIO_23] = { 1415 .type = HDA_FIXUP_FUNC, 1416 .v.func = cs420x_fixup_gpio_23, 1417 }, 1418 [CS420X_MBP101] = { 1419 .type = HDA_FIXUP_PINS, 1420 .v.pins = mbp101_pincfgs, 1421 .chained = true, 1422 .chain_id = CS420X_MBP101_COEF, 1423 }, 1424 [CS420X_MBP101_COEF] = { --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8961: Convert to direct regmap API usage
by a＠opensource.wolfsonmicro.com 13 Sep '12

13 Sep '12

From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8961.c | 426 +++++++++++++++++---------------------------- 1 file changed, 161 insertions(+), 265 deletions(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 719fb69..4ea64d6e 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -19,6 +19,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -31,277 +32,158 @@ #define WM8961_MAX_REGISTER 0xFC -static u16 wm8961_reg_defaults[] = { - 0x009F, /* R0 - Left Input volume */ - 0x009F, /* R1 - Right Input volume */ - 0x0000, /* R2 - LOUT1 volume */ - 0x0000, /* R3 - ROUT1 volume */ - 0x0020, /* R4 - Clocking1 */ - 0x0008, /* R5 - ADC & DAC Control 1 */ - 0x0000, /* R6 - ADC & DAC Control 2 */ - 0x000A, /* R7 - Audio Interface 0 */ - 0x01F4, /* R8 - Clocking2 */ - 0x0000, /* R9 - Audio Interface 1 */ - 0x00FF, /* R10 - Left DAC volume */ - 0x00FF, /* R11 - Right DAC volume */ - 0x0000, /* R12 */ - 0x0000, /* R13 */ - 0x0040, /* R14 - Audio Interface 2 */ - 0x0000, /* R15 - Software Reset */ - 0x0000, /* R16 */ - 0x007B, /* R17 - ALC1 */ - 0x0000, /* R18 - ALC2 */ - 0x0032, /* R19 - ALC3 */ - 0x0000, /* R20 - Noise Gate */ - 0x00C0, /* R21 - Left ADC volume */ - 0x00C0, /* R22 - Right ADC volume */ - 0x0120, /* R23 - Additional control(1) */ - 0x0000, /* R24 - Additional control(2) */ - 0x0000, /* R25 - Pwr Mgmt (1) */ - 0x0000, /* R26 - Pwr Mgmt (2) */ - 0x0000, /* R27 - Additional Control (3) */ - 0x0000, /* R28 - Anti-pop */ - 0x0000, /* R29 */ - 0x005F, /* R30 - Clocking 3 */ - 0x0000, /* R31 */ - 0x0000, /* R32 - ADCL signal path */ - 0x0000, /* R33 - ADCR signal path */ - 0x0000, /* R34 */ - 0x0000, /* R35 */ - 0x0000, /* R36 */ - 0x0000, /* R37 */ - 0x0000, /* R38 */ - 0x0000, /* R39 */ - 0x0000, /* R40 - LOUT2 volume */ - 0x0000, /* R41 - ROUT2 volume */ - 0x0000, /* R42 */ - 0x0000, /* R43 */ - 0x0000, /* R44 */ - 0x0000, /* R45 */ - 0x0000, /* R46 */ - 0x0000, /* R47 - Pwr Mgmt (3) */ - 0x0023, /* R48 - Additional Control (4) */ - 0x0000, /* R49 - Class D Control 1 */ - 0x0000, /* R50 */ - 0x0003, /* R51 - Class D Control 2 */ - 0x0000, /* R52 */ - 0x0000, /* R53 */ - 0x0000, /* R54 */ - 0x0000, /* R55 */ - 0x0106, /* R56 - Clocking 4 */ - 0x0000, /* R57 - DSP Sidetone 0 */ - 0x0000, /* R58 - DSP Sidetone 1 */ - 0x0000, /* R59 */ - 0x0000, /* R60 - DC Servo 0 */ - 0x0000, /* R61 - DC Servo 1 */ - 0x0000, /* R62 */ - 0x015E, /* R63 - DC Servo 3 */ - 0x0010, /* R64 */ - 0x0010, /* R65 - DC Servo 5 */ - 0x0000, /* R66 */ - 0x0001, /* R67 */ - 0x0003, /* R68 - Analogue PGA Bias */ - 0x0000, /* R69 - Analogue HP 0 */ - 0x0060, /* R70 */ - 0x01FB, /* R71 - Analogue HP 2 */ - 0x0000, /* R72 - Charge Pump 1 */ - 0x0065, /* R73 */ - 0x005F, /* R74 */ - 0x0059, /* R75 */ - 0x006B, /* R76 */ - 0x0038, /* R77 */ - 0x000C, /* R78 */ - 0x000A, /* R79 */ - 0x006B, /* R80 */ - 0x0000, /* R81 */ - 0x0000, /* R82 - Charge Pump B */ - 0x0087, /* R83 */ - 0x0000, /* R84 */ - 0x005C, /* R85 */ - 0x0000, /* R86 */ - 0x0000, /* R87 - Write Sequencer 1 */ - 0x0000, /* R88 - Write Sequencer 2 */ - 0x0000, /* R89 - Write Sequencer 3 */ - 0x0000, /* R90 - Write Sequencer 4 */ - 0x0000, /* R91 - Write Sequencer 5 */ - 0x0000, /* R92 - Write Sequencer 6 */ - 0x0000, /* R93 - Write Sequencer 7 */ - 0x0000, /* R94 */ - 0x0000, /* R95 */ - 0x0000, /* R96 */ - 0x0000, /* R97 */ - 0x0000, /* R98 */ - 0x0000, /* R99 */ - 0x0000, /* R100 */ - 0x0000, /* R101 */ - 0x0000, /* R102 */ - 0x0000, /* R103 */ - 0x0000, /* R104 */ - 0x0000, /* R105 */ - 0x0000, /* R106 */ - 0x0000, /* R107 */ - 0x0000, /* R108 */ - 0x0000, /* R109 */ - 0x0000, /* R110 */ - 0x0000, /* R111 */ - 0x0000, /* R112 */ - 0x0000, /* R113 */ - 0x0000, /* R114 */ - 0x0000, /* R115 */ - 0x0000, /* R116 */ - 0x0000, /* R117 */ - 0x0000, /* R118 */ - 0x0000, /* R119 */ - 0x0000, /* R120 */ - 0x0000, /* R121 */ - 0x0000, /* R122 */ - 0x0000, /* R123 */ - 0x0000, /* R124 */ - 0x0000, /* R125 */ - 0x0000, /* R126 */ - 0x0000, /* R127 */ - 0x0000, /* R128 */ - 0x0000, /* R129 */ - 0x0000, /* R130 */ - 0x0000, /* R131 */ - 0x0000, /* R132 */ - 0x0000, /* R133 */ - 0x0000, /* R134 */ - 0x0000, /* R135 */ - 0x0000, /* R136 */ - 0x0000, /* R137 */ - 0x0000, /* R138 */ - 0x0000, /* R139 */ - 0x0000, /* R140 */ - 0x0000, /* R141 */ - 0x0000, /* R142 */ - 0x0000, /* R143 */ - 0x0000, /* R144 */ - 0x0000, /* R145 */ - 0x0000, /* R146 */ - 0x0000, /* R147 */ - 0x0000, /* R148 */ - 0x0000, /* R149 */ - 0x0000, /* R150 */ - 0x0000, /* R151 */ - 0x0000, /* R152 */ - 0x0000, /* R153 */ - 0x0000, /* R154 */ - 0x0000, /* R155 */ - 0x0000, /* R156 */ - 0x0000, /* R157 */ - 0x0000, /* R158 */ - 0x0000, /* R159 */ - 0x0000, /* R160 */ - 0x0000, /* R161 */ - 0x0000, /* R162 */ - 0x0000, /* R163 */ - 0x0000, /* R164 */ - 0x0000, /* R165 */ - 0x0000, /* R166 */ - 0x0000, /* R167 */ - 0x0000, /* R168 */ - 0x0000, /* R169 */ - 0x0000, /* R170 */ - 0x0000, /* R171 */ - 0x0000, /* R172 */ - 0x0000, /* R173 */ - 0x0000, /* R174 */ - 0x0000, /* R175 */ - 0x0000, /* R176 */ - 0x0000, /* R177 */ - 0x0000, /* R178 */ - 0x0000, /* R179 */ - 0x0000, /* R180 */ - 0x0000, /* R181 */ - 0x0000, /* R182 */ - 0x0000, /* R183 */ - 0x0000, /* R184 */ - 0x0000, /* R185 */ - 0x0000, /* R186 */ - 0x0000, /* R187 */ - 0x0000, /* R188 */ - 0x0000, /* R189 */ - 0x0000, /* R190 */ - 0x0000, /* R191 */ - 0x0000, /* R192 */ - 0x0000, /* R193 */ - 0x0000, /* R194 */ - 0x0000, /* R195 */ - 0x0030, /* R196 */ - 0x0006, /* R197 */ - 0x0000, /* R198 */ - 0x0060, /* R199 */ - 0x0000, /* R200 */ - 0x003F, /* R201 */ - 0x0000, /* R202 */ - 0x0000, /* R203 */ - 0x0000, /* R204 */ - 0x0001, /* R205 */ - 0x0000, /* R206 */ - 0x0181, /* R207 */ - 0x0005, /* R208 */ - 0x0008, /* R209 */ - 0x0008, /* R210 */ - 0x0000, /* R211 */ - 0x013B, /* R212 */ - 0x0000, /* R213 */ - 0x0000, /* R214 */ - 0x0000, /* R215 */ - 0x0000, /* R216 */ - 0x0070, /* R217 */ - 0x0000, /* R218 */ - 0x0000, /* R219 */ - 0x0000, /* R220 */ - 0x0000, /* R221 */ - 0x0000, /* R222 */ - 0x0003, /* R223 */ - 0x0000, /* R224 */ - 0x0000, /* R225 */ - 0x0001, /* R226 */ - 0x0008, /* R227 */ - 0x0000, /* R228 */ - 0x0000, /* R229 */ - 0x0000, /* R230 */ - 0x0000, /* R231 */ - 0x0004, /* R232 */ - 0x0000, /* R233 */ - 0x0000, /* R234 */ - 0x0000, /* R235 */ - 0x0000, /* R236 */ - 0x0000, /* R237 */ - 0x0080, /* R238 */ - 0x0000, /* R239 */ - 0x0000, /* R240 */ - 0x0000, /* R241 */ - 0x0000, /* R242 */ - 0x0000, /* R243 */ - 0x0000, /* R244 */ - 0x0052, /* R245 */ - 0x0110, /* R246 */ - 0x0040, /* R247 */ - 0x0000, /* R248 */ - 0x0030, /* R249 */ - 0x0000, /* R250 */ - 0x0000, /* R251 */ - 0x0001, /* R252 - General test 1 */ +static const struct reg_default wm8961_reg_defaults[] = { + { 0, 0x009F }, /* R0 - Left Input volume */ + { 1, 0x009F }, /* R1 - Right Input volume */ + { 2, 0x0000 }, /* R2 - LOUT1 volume */ + { 3, 0x0000 }, /* R3 - ROUT1 volume */ + { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0008 }, /* R5 - ADC & DAC Control 1 */ + { 6, 0x0000 }, /* R6 - ADC & DAC Control 2 */ + { 7, 0x000A }, /* R7 - Audio Interface 0 */ + { 8, 0x01F4 }, /* R8 - Clocking2 */ + { 9, 0x0000 }, /* R9 - Audio Interface 1 */ + { 10, 0x00FF }, /* R10 - Left DAC volume */ + { 11, 0x00FF }, /* R11 - Right DAC volume */ + + { 14, 0x0040 }, /* R14 - Audio Interface 2 */ + + { 17, 0x007B }, /* R17 - ALC1 */ + { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x0032 }, /* R19 - ALC3 */ + { 20, 0x0000 }, /* R20 - Noise Gate */ + { 21, 0x00C0 }, /* R21 - Left ADC volume */ + { 22, 0x00C0 }, /* R22 - Right ADC volume */ + { 23, 0x0120 }, /* R23 - Additional control(1) */ + { 24, 0x0000 }, /* R24 - Additional control(2) */ + { 25, 0x0000 }, /* R25 - Pwr Mgmt (1) */ + { 26, 0x0000 }, /* R26 - Pwr Mgmt (2) */ + { 27, 0x0000 }, /* R27 - Additional Control (3) */ + { 28, 0x0000 }, /* R28 - Anti-pop */ + + { 30, 0x005F }, /* R30 - Clocking 3 */ + + { 32, 0x0000 }, /* R32 - ADCL signal path */ + { 33, 0x0000 }, /* R33 - ADCR signal path */ + + { 40, 0x0000 }, /* R40 - LOUT2 volume */ + { 41, 0x0000 }, /* R41 - ROUT2 volume */ + + { 47, 0x0000 }, /* R47 - Pwr Mgmt (3) */ + { 48, 0x0023 }, /* R48 - Additional Control (4) */ + { 49, 0x0000 }, /* R49 - Class D Control 1 */ + + { 51, 0x0003 }, /* R51 - Class D Control 2 */ + + { 56, 0x0106 }, /* R56 - Clocking 4 */ + { 57, 0x0000 }, /* R57 - DSP Sidetone 0 */ + { 58, 0x0000 }, /* R58 - DSP Sidetone 1 */ + + { 60, 0x0000 }, /* R60 - DC Servo 0 */ + { 61, 0x0000 }, /* R61 - DC Servo 1 */ + + { 63, 0x015E }, /* R63 - DC Servo 3 */ + + { 65, 0x0010 }, /* R65 - DC Servo 5 */ + + { 68, 0x0003 }, /* R68 - Analogue PGA Bias */ + { 69, 0x0000 }, /* R69 - Analogue HP 0 */ + + { 71, 0x01FB }, /* R71 - Analogue HP 2 */ + { 72, 0x0000 }, /* R72 - Charge Pump 1 */ + + { 82, 0x0000 }, /* R82 - Charge Pump B */ + + { 87, 0x0000 }, /* R87 - Write Sequencer 1 */ + { 88, 0x0000 }, /* R88 - Write Sequencer 2 */ + { 89, 0x0000 }, /* R89 - Write Sequencer 3 */ + { 90, 0x0000 }, /* R90 - Write Sequencer 4 */ + { 91, 0x0000 }, /* R91 - Write Sequencer 5 */ + { 92, 0x0000 }, /* R92 - Write Sequencer 6 */ + { 93, 0x0000 }, /* R93 - Write Sequencer 7 */ + + { 252, 0x0001 }, /* R252 - General test 1 */ }; struct wm8961_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk; }; -static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8961_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM8961_SOFTWARE_RESET: case WM8961_WRITE_SEQUENCER_7: case WM8961_DC_SERVO_1: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm8961_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8961_LEFT_INPUT_VOLUME: + case WM8961_RIGHT_INPUT_VOLUME: + case WM8961_LOUT1_VOLUME: + case WM8961_ROUT1_VOLUME: + case WM8961_CLOCKING1: + case WM8961_ADC_DAC_CONTROL_1: + case WM8961_ADC_DAC_CONTROL_2: + case WM8961_AUDIO_INTERFACE_0: + case WM8961_CLOCKING2: + case WM8961_AUDIO_INTERFACE_1: + case WM8961_LEFT_DAC_VOLUME: + case WM8961_RIGHT_DAC_VOLUME: + case WM8961_AUDIO_INTERFACE_2: + case WM8961_SOFTWARE_RESET: + case WM8961_ALC1: + case WM8961_ALC2: + case WM8961_ALC3: + case WM8961_NOISE_GATE: + case WM8961_LEFT_ADC_VOLUME: + case WM8961_RIGHT_ADC_VOLUME: + case WM8961_ADDITIONAL_CONTROL_1: + case WM8961_ADDITIONAL_CONTROL_2: + case WM8961_PWR_MGMT_1: + case WM8961_PWR_MGMT_2: + case WM8961_ADDITIONAL_CONTROL_3: + case WM8961_ANTI_POP: + case WM8961_CLOCKING_3: + case WM8961_ADCL_SIGNAL_PATH: + case WM8961_ADCR_SIGNAL_PATH: + case WM8961_LOUT2_VOLUME: + case WM8961_ROUT2_VOLUME: + case WM8961_PWR_MGMT_3: + case WM8961_ADDITIONAL_CONTROL_4: + case WM8961_CLASS_D_CONTROL_1: + case WM8961_CLASS_D_CONTROL_2: + case WM8961_CLOCKING_4: + case WM8961_DSP_SIDETONE_0: + case WM8961_DSP_SIDETONE_1: + case WM8961_DC_SERVO_0: + case WM8961_DC_SERVO_1: + case WM8961_DC_SERVO_3: + case WM8961_DC_SERVO_5: + case WM8961_ANALOGUE_PGA_BIAS: + case WM8961_ANALOGUE_HP_0: + case WM8961_ANALOGUE_HP_2: + case WM8961_CHARGE_PUMP_1: + case WM8961_CHARGE_PUMP_B: + case WM8961_WRITE_SEQUENCER_1: + case WM8961_WRITE_SEQUENCER_2: + case WM8961_WRITE_SEQUENCER_3: + case WM8961_WRITE_SEQUENCER_4: + case WM8961_WRITE_SEQUENCER_5: + case WM8961_WRITE_SEQUENCER_6: + case WM8961_WRITE_SEQUENCER_7: + case WM8961_GENERAL_TEST_1: + return true; + default: + return false; } } @@ -958,11 +840,12 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { + struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -975,9 +858,9 @@ static int wm8961_probe(struct snd_soc_codec *codec) } /* This isn't volatile - readback doesn't correspond to write */ - codec->cache_bypass = 1; + regcache_cache_bypass(wm8961->regmap, true); reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); - codec->cache_bypass = 0; + regcache_cache_bypass(wm8961->regmap, false); dev_info(codec->dev, "WM8961 family %d revision %c\n", (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) @@ -1066,10 +949,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .suspend = wm8961_suspend, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8961_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8961_reg_defaults, - .volatile_register = wm8961_volatile_register, +}; + +static const struct regmap_config wm8961_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = WM8961_MAX_REGISTER, + + .reg_defaults = wm8961_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8961_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8961_volatile, + .readable_reg = wm8961_readable, }; static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, @@ -1083,6 +975,10 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, if (wm8961 == NULL) return -ENOMEM; + wm8961->regmap = devm_regmap_init_i2c(i2c, &wm8961_regmap); + if (IS_ERR(wm8961->regmap)) + return PTR_ERR(wm8961->regmap); + i2c_set_clientdata(i2c, wm8961); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.10.4

1 1

Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa-mixer: Don't use the mono path in the analog-stereo mapping.
by Raymond Yau 13 Sep '12

13 Sep '12

2012-9-12 下午9:04 於 "Tanu Kaskinen" <tanuk(a)iki.fi> 寫道： > > As far as I can see, having a mono path in a stereo mapping doesn't > make any sense. It also causes breakage: if the Master Mono mixer > element has two volume channels, Because ac97 codecs have an analog mixer which can downmix stereo to the mono out pin. Some hda codecs also have the mono out pin which connected to a mixer widget for downmixing stereo from stereo DAC to mono The master mono playback volume control of ac97 has only one channel , so it is a bug of vm 's ac97 codec register emulation if the control has two volume channels Either removed the master mono register emulation from vm or pulseaudio ignore the master mono switch of vm's ac97 codec > the analog-output path gets removed > due to being a subset of analog-output-mono, and that in turn causes > the Master element getting muted. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… The above patch try to assign the headphone path for the USB headsets which have speaker playback volume control since speaker playback volume control is not in pulseaudio ' s headphone path. >Users generally don't like that. > > BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=54673 It look like a bug in output volume calculation of ac97 codec emulation in those virtual machine if the emulated ac97 codec only has 16 bits DAC resolution or pulseaudio does not unmute pcm playback switch on those ac97 codec after user mute the sound in sound preference and pulseaudio mute both master playback switch and pcm playback switch Some tablet with alc203 use the master mono output for the mono speaker http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=8f4f4ef6fed55… For als4000 the master mono is only enum control to change the channel map of the mono out Simple mixer control 'Master Mono',0 Capabilities: pswitch pswitch-joined cenum Items: 'L chan only' 'R chan only' 'L ch/2 + R ch/2' Item0: 'L chan only' https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/827393

1 0

Re: [alsa-devel] [PATCH 8/9] sound/soc/soc-core.c: Remove useless kfree
by Mark Brown 13 Sep '12

13 Sep '12

On Wed, Sep 12, 2012 at 05:06:46PM +0200, Peter Senna Tschudin wrote: > From: Peter Senna Tschudin <peter.senna(a)gmail.com> > > Remove useless kfree() and clean up code related to the removal. > > The semantic patch that finds this problem is as follows: > (http://coccinelle.lip6.fr/) Applied, thanks.

1 0

[alsa-devel] [RFC PATCH 0/5] mpc5200 asoc fixups
by Eric Millbrandt 13 Sep '12

13 Sep '12

The mpc5200 asoc code did not run after the multi-codec patches. This series fixes the existing code and adds a generic platform driver that allows the audio layout to be described in the device tree. Eric Millbrandt (5): ASoC: fsl: mpc5200 multi-codec fixups ASoC: fsl: mpc5200 combine psc_dma platform data ASoC: fsl: mpc5200-soc-audio driver arch/powerpc/boot/dts pcm030 add mpc5200-soc-audio node ASoC: fsl: mpc5200 remove pcm030 and efika audio fabric .../devicetree/bindings/powerpc/fsl/mpc5200.txt | 17 ++ arch/powerpc/boot/dts/pcm030.dts | 23 ++- arch/powerpc/platforms/52xx/Kconfig | 1 + sound/soc/fsl/Kconfig | 20 +-- sound/soc/fsl/Makefile | 3 +- sound/soc/fsl/efika-audio-fabric.c | 91 -------- sound/soc/fsl/mpc5200_dma.c | 28 +-- sound/soc/fsl/mpc5200_dma.h | 3 + sound/soc/fsl/mpc5200_psc_ac97.c | 13 +- sound/soc/fsl/mpc5200_psc_i2s.c | 8 + sound/soc/fsl/mpc5200_soc_audio.c | 232 ++++++++++++++++++++ sound/soc/fsl/pcm030-audio-fabric.c | 91 -------- 12 files changed, 306 insertions(+), 224 deletions(-) delete mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_soc_audio.c delete mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- 1.7.2.5

3 16

[alsa-devel] [PATCH 1/2] [v2] ASoC: fsl: use snd_soc_register_card to register the card
by Timur Tabi 13 Sep '12

13 Sep '12

Use snd_soc_register_card() instead of platform_device_alloc("soc-audio") to register the sound card from the machine drivers. Although platform_device_alloc is officially deprecated, it is still used by several other drivers. Unfortunately, something is broken somewhere and the Freescale drivers don't register properly when using it. Since we need to transition to snd_soc_register_card() anyway, this fixes the problem and updates our code at the same time. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 23 +++++------------------ sound/soc/fsl/p1022_ds.c | 22 ++++------------------ 2 files changed, 9 insertions(+), 36 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 60bcba1..5479882 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -192,7 +192,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; int ret = -ENODEV; const char *sprop; @@ -341,34 +340,22 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->card.probe = mpc8610_hpcd_machine_probe; machine_data->card.remove = mpc8610_hpcd_machine_remove; machine_data->card.name = pdev->name; /* The platform driver name */ + machine_data->card.owner = THIS_MODULE; + machine_data->card.dev = &pdev->dev; machine_data->card.num_links = 2; machine_data->card.dai_link = machine_data->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &machine_data->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&machine_data->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error_sound; + dev_err(&pdev->dev, "could not register card\n"); + goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; -error_sound: - platform_device_put(sound_device); error: kfree(machine_data); error_alloc: diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 50adf40..a44a8c3 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -202,7 +202,6 @@ static int p1022_ds_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct machine_data *mdata; int ret = -ENODEV; const char *sprop; @@ -349,36 +348,23 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->card.probe = p1022_ds_machine_probe; mdata->card.remove = p1022_ds_machine_remove; mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.owner = THIS_MODULE; + mdata->card.dev = &pdev->dev; mdata->card.num_links = 2; mdata->card.dai_link = mdata->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &mdata->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&mdata->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; error: - if (sound_device) - platform_device_put(sound_device); - kfree(mdata); error_put: of_node_put(codec_np); -- 1.7.3.4

2 5

[alsa-devel] [PATCH] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug unconditionally
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_enable_hdmi(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_disable_hdmi(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 5d02aad..d39ed58 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -676,8 +676,7 @@ static void intel_disable_hdmi(struct intel_encoder *encoder) u32 temp; u32 enable_bits = SDVO_ENABLE; - if (intel_hdmi->has_audio) - enable_bits |= SDVO_AUDIO_ENABLE; + enable_bits |= SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

3 3

[alsa-devel] [PATCH] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_hdmi_dpms(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_hdmi_dpms(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 98f6024..12dc330 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -609,7 +609,7 @@ static void intel_hdmi_dpms(struct drm_encoder *encoder, int mode) u32 temp; u32 enable_bits = SDVO_ENABLE; - if (intel_hdmi->has_audio) + if (intel_hdmi->has_audio || mode != DRM_MODE_DPMS_ON) enable_bits |= SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: core: Mark regmap CODEC register maps as dirty when suspending
by Mark Brown 12 Sep '12

12 Sep '12

The core has for a long time had support for marking the register maps of devices dirty when suspending so that they are resynced on resume. Also implement this feature for CODECs using regmap. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0c7e477..b94ce32 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -635,6 +635,8 @@ int snd_soc_suspend(struct device *dev) codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; + if (codec->using_regmap) + regcache_mark_dirty(codec->control_data); break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH - snd_ice1724 1/1] Use linear scale for AK4396 volume control.
by Takashi Iwai 12 Sep '12

12 Sep '12

At Wed, 12 Sep 2012 10:12:06 -0400, athena(a)fftw.org wrote: > > From: Matteo Frigo <athena(a)fftw.org> > > The AK4396 DAC has a linear-scale attentuator, but > sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is > not quite right. This patch restores the correct scale, borrowing > from the ak4396 code in sound/pci/oxygen/oxygen.c. > > Signed-off-by: Matteo Frigo <athena(a)fftw.org> Applied now. Thanks. Takashi > > diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c > index 7f2b63f..7bf093c 100644 > --- a/sound/pci/ice1712/prodigy_hifi.c > +++ b/sound/pci/ice1712/prodigy_hifi.c > @@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem > } > > static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); > +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > > static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { > { > @@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { > .info = ak4396_dac_vol_info, > .get = ak4396_dac_vol_get, > .put = ak4396_dac_vol_put, > - .tlv = { .p = db_scale_wm_dac }, > + .tlv = { .p = ak4396_db_scale }, > }, > }; > > -- > 1.7.10.4 >

1 0

[alsa-devel] [PATCH] ASoC: wm8776: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 75 ++++++++++++++++++++++++++++++++++++--------- 1 file changed, 60 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 879c356..c32249d 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -19,6 +19,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/of_device.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <sound/core.h> @@ -37,18 +38,46 @@ enum wm8776_chip_type { /* codec private data */ struct wm8776_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk[2]; }; -static const u16 wm8776_reg[WM8776_CACHEREGNUM] = { - 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */ - 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */ - 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */ - 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */ - 0xa6, 0x01, 0x01 +static const struct reg_default wm8776_reg_defaults[] = { + { 0, 0x79 }, + { 1, 0x79 }, + { 2, 0x79 }, + { 3, 0xff }, + { 4, 0xff }, + { 5, 0xff }, + { 6, 0x00 }, + { 7, 0x90 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x22 }, + { 11, 0x22 }, + { 12, 0x22 }, + { 13, 0x08 }, + { 14, 0xcf }, + { 15, 0xcf }, + { 16, 0x7b }, + { 17, 0x00 }, + { 18, 0x32 }, + { 19, 0x00 }, + { 20, 0xa6 }, + { 21, 0x01 }, + { 22, 0x01 }, }; +static bool wm8776_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8776_RESET: + return true; + default: + return false; + } +} + static int wm8776_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8776_RESET, 0); @@ -306,6 +335,8 @@ static int wm8776_set_sysclk(struct snd_soc_dai *dai, static int wm8776_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -313,7 +344,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8776->regmap); /* Disable the global powerdown; DAPM does the rest */ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); @@ -396,10 +427,9 @@ static int wm8776_resume(struct snd_soc_codec *codec) static int wm8776_probe(struct snd_soc_codec *codec) { - struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -434,9 +464,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .suspend = wm8776_suspend, .resume = wm8776_resume, .set_bias_level = wm8776_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8776_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8776_reg, .controls = wm8776_snd_controls, .num_controls = ARRAY_SIZE(wm8776_snd_controls), @@ -452,6 +479,18 @@ static const struct of_device_id wm8776_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8776_of_match); +static const struct regmap_config wm8776_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8776_RESET, + + .reg_defaults = wm8776_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8776_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8776_volatile, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8776_spi_probe(struct spi_device *spi) { @@ -463,7 +502,10 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) if (wm8776 == NULL) return -ENOMEM; - wm8776->control_type = SND_SOC_SPI; + wm8776->regmap = devm_regmap_init_spi(spi, &wm8776_regmap); + if (IS_ERR(wm8776->regmap)) + return PTR_ERR(wm8776->regmap); + spi_set_drvdata(spi, wm8776); ret = snd_soc_register_codec(&spi->dev, @@ -501,8 +543,11 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, if (wm8776 == NULL) return -ENOMEM; + wm8776->regmap = devm_regmap_init_i2c(i2c, &wm8776_regmap); + if (IS_ERR(wm8776->regmap)) + return PTR_ERR(wm8776->regmap); + i2c_set_clientdata(i2c, wm8776); - wm8776->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); -- 1.7.10.4

1 0

[alsa-devel] ASOC: dapm event errors not reported
by Pierre Peiffer 12 Sep '12

12 Sep '12

Hi, I have observed that in dapm, most of the errors of widget event calls are reported in kernel log but not returned to the caller. This leads in some cases to a situation where the error is not reported to the userspace caller, letting it continue, even if it can't and following calls from userspace will fail as well. Is there a reason for this ? Thanks & BR -- Pierre

1 0

[alsa-devel] Amixer: suppressing errors when called from stdin
by Henrik Fasth 12 Sep '12

12 Sep '12

Hi, We have some problems with amixer always returning 0 when called from stdin (exec_stdin). The global variable ignore_error is always set to 1 in function exec_stdin which leads to sset and cset always returning 0 even if an error occured. Are there any specific reasons for suppressing error messages and error return values like this? Br Henrik

1 0

[alsa-devel] [PATCH V3] ASoC: Davinci: machine: Add device tree binding
by Hebbar, Gururaja 12 Sep '12

12 Sep '12

Device tree support for Davinci Machine driver When the board boots with device tree, the driver will receive card, codec, dai interface details (like the card name, DAPM routing map, phandle for the audio components described in the dts file, codec mclk speed). The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. When here, code indentation and comment style is also fixed Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- Changes from V1: - Change DT parameter from "ti,codec-clock" to "ti,codec-clock-rate" - add more explanation to DT parameters Changes from V2: - Update Documentation details - Remove irrelevant commit message :000000 100644 0000000... b248014... A Documentation/devicetree/bindings/sound/davinci-evm-audio.txt :100644 100644 ab0ad45... ca2a547... M sound/soc/davinci/davinci-evm.c .../bindings/sound/davinci-evm-audio.txt | 66 +++++++ sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++-- 2 files changed, 231 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 0000000..b248014 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -0,0 +1,66 @@ +* Texas Instruments SoC audio setups with TLV320AIC3X Codec + +Required properties: +- compatible : "ti,davinci-evm-audio" +- ti,model : The user-visible name of this sound complex. +- ti,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 for sources and + sinks are the codec's pins, and the jacks on the board: + + Codec pins: + + * MIC3L + * MIC3R + * LINE1L + * LINE2L + * LINE1R + * LINE2R + + Board connectors: + + * Headphone Jack + * Line Out + * Mic Jack + +- ti,mcasp-controller : The phandle of the McASP controller +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec + +- dai-data : A list of DAI data used by SOC code to register + DAI, Codecs platform. + The string index "should" be as shown below. +dai-data = +"<DAI Name>", "<DAI Stream Name>", +"<CODEC DAI Name>", "<true if evm_aic3x_init is required, else false>", +"<evm ops required (evm_ops or evm_spdif_ops)>"; + +Here fields +"<DAI Name>" : used to indicate the DAI Name +"<DAI Stream Name>" : used to indicate the Stream Name +"<CODEC DAI Name>" : used to bind the link between Codec DAI and ASOC DAI + +Machine related options +"<true/false>" : Whether the machine specific initialization + : evm_aic3x_init() is required + +"<evm ops required>" : Which hardware ops function is to be used. + : (evm_ops or evm_spdif_ops) + : use evm-spdif-ops if DAI is working in DIT mode + : else use evm-ops. These ops setup hw param callbacks + : which are used to setup CODEC/cpu DAI configuration + : and codec system clock. + +Example: + +sound { + compatible = "ti,davinci-evm-audio"; + ti,model = "DA830 EVM"; + ti,audio-codec = <&tlv320aic3x>; + ti,mcasp-controller = <&mcasp1>; + ti,codec-clock-rate = <12000000>; + dai-data = + "TLV320AIC3X", "AIC3X", + "tlv320aic3x-hifi", "true", + "evm-ops"; +}; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index ab0ad45..ca2a547 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -34,27 +34,38 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *soc_card = codec->card; + struct device_node *np = soc_card->dev->of_node; int ret = 0; unsigned sysclk; - /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || - machine_is_davinci_dm365_evm()) - sysclk = 27000000; - - /* ASP0 in DM6446 EVM is clocked by U55, as configured by - * board-dm644x-evm.c using GPIOs from U18. There are six - * options; here we "know" we use a 48 KHz sample rate. - */ - else if (machine_is_davinci_evm()) - sysclk = 12288000; - - else if (machine_is_davinci_da830_evm() || - machine_is_davinci_da850_evm()) - sysclk = 24576000; - - else - return -EINVAL; + if (np) { + ret = of_property_read_u32(np, "ti,codec-clock-rate", &sysclk); + if (ret < 0) + return ret; + } else { + /* ASP1 on DM355 EVM is clocked by an external oscillator */ + if (machine_is_davinci_dm355_evm() || + machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) + sysclk = 27000000; + + /* + * ASP0 in DM6446 EVM is clocked by U55, as configured by + * board-dm644x-evm.c using GPIOs from U18. There are six + * options; here we "know" we use a 48 KHz sample rate. + */ + else if (machine_is_davinci_evm()) + sysclk = 12288000; + + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) + sysclk = 24576000; + + else + return -EINVAL; + } /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); @@ -127,13 +138,22 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + int ret; /* Add davinci-evm specific widgets */ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + if (np) { + ret = snd_soc_of_parse_audio_routing(codec->card, + "ti,audio-routing"); + if (ret) + return ret; + } else { + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + } /* not connected */ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); @@ -282,6 +302,113 @@ static struct snd_soc_card da850_snd_soc_card = { .num_links = 1, }; +#if defined(CONFIG_OF) +/* + * This struct is just used as place holder. It will be filled with + * data from dt node + */ +static struct snd_soc_dai_link evm_dai = { +}; + +/* davinci evm audio machine driver */ +static struct snd_soc_card evm_soc_card = { + .owner = THIS_MODULE, + .dai_link = &evm_dai, + .num_links = 1, +}; + +static int davinci_evm_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + const char *stringPtr, *propname; + u32 val; + int ret = 0; + + propname = "dai-data"; + val = of_property_count_strings(np, propname); + if (val < 0) + return val; + + ret = of_property_read_string_index(np, propname, 0, &evm_dai.name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 1, + &evm_dai.stream_name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 2, + &evm_dai.codec_dai_name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 3, &stringPtr); + if (ret < 0) + return ret; + else + if (strcasecmp(stringPtr, "true") == 0) + evm_dai.init = evm_aic3x_init; + + ret = of_property_read_string_index(np, propname, 4, &stringPtr); + if (ret < 0) { + return ret; + } else { + if (strcasecmp(stringPtr, "evm-ops") == 0) + evm_dai.ops = &evm_ops; + else if (strcasecmp(stringPtr, "evm-spdif-ops") == 0) + evm_dai.ops = &evm_spdif_ops; + } + + evm_dai.codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0); + if (!evm_dai.codec_of_node) + return -EINVAL; + + evm_dai.cpu_dai_of_node = of_parse_phandle(np, + "ti,mcasp-controller", 0); + if (!evm_dai.cpu_dai_of_node) + return -EINVAL; + + evm_dai.platform_of_node = evm_dai.cpu_dai_of_node; + + evm_soc_card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); + if (ret) + return ret; + + ret = snd_soc_register_card(&evm_soc_card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + + return ret; +} + +static int __devexit davinci_evm_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id davinci_evm_dt_ids[] = { + { .compatible = "ti,davinci-evm-audio", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, davinci_mcasp_dt_ids); + +static struct platform_driver davinci_evm_driver = { + .probe = davinci_evm_probe, + .remove = __devexit_p(davinci_evm_remove), + .driver = { + .name = "davinci_evm", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(davinci_evm_dt_ids), + }, +}; +#endif + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -289,6 +416,14 @@ static int __init evm_init(void) struct snd_soc_card *evm_snd_dev_data; int index; int ret; +#if defined(CONFIG_OF) + struct device_node *np; + + np = of_find_compatible_node(NULL, NULL, "ti,davinci-evm-audio"); + if (np) { + return platform_driver_register(&davinci_evm_driver); + } +#endif if (machine_is_davinci_evm()) { evm_snd_dev_data = &dm6446_snd_soc_card_evm; @@ -325,6 +460,16 @@ static int __init evm_init(void) static void __exit evm_exit(void) { +#if defined(CONFIG_OF) + struct device_node *np; + + np = of_find_compatible_node(NULL, NULL, "ti,davinci-evm-audio"); + if (np) { + platform_driver_unregister(&davinci_evm_driver); + return; + } +#endif + platform_device_unregister(evm_snd_device); } -- 1.7.1

2 6

[alsa-devel] [PATCH] ALSA: hda_intel: add position_fix quirk for Asus K53E
by Catalin Iacob 12 Sep '12

12 Sep '12

Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob <iacobcatalin(a)gmail.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c6..228cdf9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2701,6 +2701,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), -- 1.7.11.5

2 4

[alsa-devel] [PATCH 1/2] ASoC: wm8737: Move regulator acquisition to device registration
by Mark Brown 12 Sep '12

12 Sep '12

This is better style as we acquire resources we will need before we go into the ASoC card probe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8737.c | 41 +++++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 8c25442..3bf5bc7 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -557,7 +557,7 @@ static int wm8737_resume(struct snd_soc_codec *codec) static int wm8737_probe(struct snd_soc_codec *codec) { struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - int ret, i; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8737->control_type); if (ret != 0) { @@ -565,16 +565,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) - wm8737->supplies[i].supply = wm8737_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8737->supplies), - wm8737->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); if (ret != 0) { @@ -607,17 +597,12 @@ static int wm8737_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); - return ret; } static int wm8737_remove(struct snd_soc_codec *codec) { - struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); return 0; } @@ -645,13 +630,23 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&i2c->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8737); wm8737->control_type = SND_SOC_I2C; @@ -691,13 +686,23 @@ static struct i2c_driver wm8737_i2c_driver = { static int __devinit wm8737_spi_probe(struct spi_device *spi) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&spi->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&spi->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + wm8737->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8737); -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm8728: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8728.c | 45 ++++++++++++++++++++++++++++----------------- 1 file changed, 28 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f274c25..00a12a0 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -17,6 +17,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -35,16 +36,16 @@ * the volume update bits, mute the output and enable infinite zero * detect. */ -static const u16 wm8728_reg_defaults[] = { - 0x1ff, - 0x1ff, - 0x001, - 0x100, +static const struct reg_default wm8728_reg_defaults[] = { + { 0, 0x1ff }, + { 1, 0x1ff }, + { 2, 0x001 }, + { 3, 0x100 }, }; /* codec private data */ struct wm8728_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); @@ -162,8 +163,8 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8728_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); u16 reg; - int i; switch (level) { case SND_SOC_BIAS_ON: @@ -175,9 +176,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); /* ..then sync in the register cache. */ - for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) - snd_soc_write(codec, i, - snd_soc_read(codec, i)); + regcache_sync(wm8728->regmap); } break; @@ -229,10 +228,9 @@ static int wm8728_resume(struct snd_soc_codec *codec) static int wm8728_probe(struct snd_soc_codec *codec) { - struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8728->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", ret); @@ -257,9 +255,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .suspend = wm8728_suspend, .resume = wm8728_resume, .set_bias_level = wm8728_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8728_reg_defaults, .controls = wm8728_snd_controls, .num_controls = ARRAY_SIZE(wm8728_snd_controls), .dapm_widgets = wm8728_dapm_widgets, @@ -274,6 +269,16 @@ static const struct of_device_id wm8728_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8728_of_match); +static const struct regmap_config wm8728_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8728_IFCTL, + + .reg_defaults = wm8728_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8728_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8728_spi_probe(struct spi_device *spi) { @@ -285,7 +290,10 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi) if (wm8728 == NULL) return -ENOMEM; - wm8728->control_type = SND_SOC_SPI; + wm8728->regmap = devm_regmap_init_spi(spi, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + spi_set_drvdata(spi, wm8728); ret = snd_soc_register_codec(&spi->dev, @@ -324,8 +332,11 @@ static __devinit int wm8728_i2c_probe(struct i2c_client *i2c, if (wm8728 == NULL) return -ENOMEM; + wm8728->regmap = devm_regmap_init_i2c(i2c, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + i2c_set_clientdata(i2c, wm8728); - wm8728->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8728, &wm8728_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8711: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8711.c | 38 ++++++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 1b1209a..5c165d0 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -32,7 +33,7 @@ /* codec private data */ struct wm8711_priv { - enum snd_soc_control_type bus_type; + struct regmap *regmap; unsigned int sysclk; }; @@ -42,9 +43,9 @@ struct wm8711_priv { * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { - 0x0079, 0x0079, 0x000a, 0x0008, - 0x009f, 0x000a, 0x0000, 0x0000 +static const struct reg_default wm8711_reg_defaults[] = { + { 0, 0x0079 }, { 1, 0x0079 }, { 2, 0x000a }, { 3, 0x0008 }, + { 4, 0x009f }, { 5, 0x000a }, { 6, 0x0000 }, { 7, 0x0000 }, }; #define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) @@ -289,6 +290,7 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8711_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f; switch (level) { @@ -299,7 +301,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - snd_soc_cache_sync(codec); + regcache_sync(wm8711->regmap); snd_soc_write(codec, WM8711_PWR, reg | 0x0040); break; @@ -353,10 +355,9 @@ static int wm8711_resume(struct snd_soc_codec *codec) static int wm8711_probe(struct snd_soc_codec *codec) { - struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -391,9 +392,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .suspend = wm8711_suspend, .resume = wm8711_resume, .set_bias_level = wm8711_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8711_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8711_reg, .controls = wm8711_snd_controls, .num_controls = ARRAY_SIZE(wm8711_snd_controls), .dapm_widgets = wm8711_dapm_widgets, @@ -408,6 +406,16 @@ static const struct of_device_id wm8711_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8711_of_match); +static const struct regmap_config wm8711_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8711_RESET, + + .reg_defaults = wm8711_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8711_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -419,8 +427,11 @@ static int __devinit wm8711_spi_probe(struct spi_device *spi) if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_spi(spi, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + spi_set_drvdata(spi, wm8711); - wm8711->bus_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); @@ -458,8 +469,11 @@ static __devinit int wm8711_i2c_probe(struct i2c_client *client, if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_i2c(client, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + i2c_set_clientdata(client, wm8711); - wm8711->bus_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8580: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 104 +++++++++++++++++++++++++++++++++------------ 1 file changed, 78 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index cc198df..2a07c70 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008-11 Wolfson Microelectronics PLC. + * Copyright 2008-12 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -23,6 +23,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -157,21 +158,60 @@ * We can't read the WM8580 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8580_reg[] = { - 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ - 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ - 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ - 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R27*/ - 0x01f0, 0x0040, 0x0000, 0x0000, /*R31(0x1F)*/ - 0x0000, 0x0000, 0x0031, 0x000b, /*R35*/ - 0x0039, 0x0000, 0x0010, 0x0032, /*R39*/ - 0x0054, 0x0076, 0x0098, 0x0000, /*R43(0x2B)*/ - 0x0000, 0x0000, 0x0000, 0x0000, /*R47*/ - 0x0000, 0x0000, 0x005e, 0x003e, /*R51(0x33)*/ - 0x0000, 0x0000 /*R53*/ +static const struct reg_default wm8580_reg_defaults[] = { + { 0, 0x0121 }, + { 1, 0x017e }, + { 2, 0x007d }, + { 3, 0x0014 }, + { 4, 0x0121 }, + { 5, 0x017e }, + { 6, 0x007d }, + { 7, 0x0194 }, + { 8, 0x0010 }, + { 9, 0x0002 }, + { 10, 0x0002 }, + { 11, 0x00c2 }, + { 12, 0x0182 }, + { 13, 0x0082 }, + { 14, 0x000a }, + { 15, 0x0024 }, + { 16, 0x0009 }, + { 17, 0x0000 }, + { 18, 0x00ff }, + { 19, 0x0000 }, + { 20, 0x00ff }, + { 21, 0x00ff }, + { 22, 0x00ff }, + { 23, 0x00ff }, + { 24, 0x00ff }, + { 25, 0x00ff }, + { 26, 0x00ff }, + { 27, 0x00ff }, + { 28, 0x01f0 }, + { 29, 0x0040 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0031 }, + { 35, 0x000b }, + { 36, 0x0039 }, + { 37, 0x0000 }, + { 38, 0x0010 }, + { 39, 0x0032 }, + { 40, 0x0054 }, + { 41, 0x0076 }, + { 42, 0x0098 }, + { 43, 0x0000 }, + { 44, 0x0000 }, + { 45, 0x0000 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0000 }, + { 50, 0x005e }, + { 51, 0x003e }, + { 52, 0x0000 }, }; struct pll_state { @@ -188,7 +228,7 @@ static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { /* codec private data */ struct wm8580_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES]; struct pll_state a; struct pll_state b; @@ -203,14 +243,16 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; + struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); unsigned int reg = mc->reg; unsigned int reg2 = mc->rreg; int ret; - /* Clear the register cache so we write without VU set */ - reg_cache[reg] = 0; - reg_cache[reg2] = 0; + /* Clear the register cache VU so we write without VU set */ + regcache_cache_only(wm8580->regmap, true); + regmap_update_bits(wm8580->regmap, reg, 0x100, 0x000); + regmap_update_bits(wm8580->regmap, reg2, 0x100, 0x000); + regcache_cache_only(wm8580->regmap, false); ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) @@ -817,7 +859,7 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0,i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8580->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -875,9 +917,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8580_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8580_reg, .controls = wm8580_snd_controls, .num_controls = ARRAY_SIZE(wm8580_snd_controls), @@ -892,6 +931,16 @@ static const struct of_device_id wm8580_of_match[] = { { }, }; +static const struct regmap_config wm8580_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8580_MAX_REGISTER, + + .reg_defaults = wm8580_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8580_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -904,8 +953,11 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, if (wm8580 == NULL) return -ENOMEM; + wm8580->regmap = devm_regmap_init_i2c(i2c, &wm8580_regmap); + if (IS_ERR(wm8580->regmap)) + return PTR_ERR(wm8580->regmap); + i2c_set_clientdata(i2c, wm8580); - wm8580->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8580, wm8580_dai, ARRAY_SIZE(wm8580_dai)); -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm8510: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 105 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 80 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 98f28a0..d9a3ca6 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -33,22 +34,63 @@ * We can't read the WM8510 register space when we are * using 2 wire for device control, so we cache them instead. */ -static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0050, 0x0000, 0x0140, 0x0000, - 0x0000, 0x0000, 0x0000, 0x00ff, - 0x0000, 0x0000, 0x0100, 0x00ff, - 0x0000, 0x0000, 0x012c, 0x002c, - 0x002c, 0x002c, 0x002c, 0x0000, - 0x0032, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0038, 0x000b, 0x0032, 0x0000, - 0x0008, 0x000c, 0x0093, 0x00e9, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0003, 0x0010, 0x0000, 0x0000, - 0x0000, 0x0002, 0x0001, 0x0000, - 0x0000, 0x0000, 0x0039, 0x0000, - 0x0001, +static const struct reg_default wm8510_reg_defaults[] = { + { 1, 0x0000 }, + { 2, 0x0000 }, + { 3, 0x0000 }, + { 4, 0x0050 }, + { 5, 0x0000 }, + { 6, 0x0140 }, + { 7, 0x0000 }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x0000 }, + { 11, 0x00ff }, + { 12, 0x0000 }, + { 13, 0x0000 }, + { 14, 0x0100 }, + { 15, 0x00ff }, + { 16, 0x0000 }, + { 17, 0x0000 }, + { 18, 0x012c }, + { 19, 0x002c }, + { 20, 0x002c }, + { 21, 0x002c }, + { 22, 0x002c }, + { 23, 0x0000 }, + { 24, 0x0032 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0038 }, + { 33, 0x000b }, + { 34, 0x0032 }, + { 35, 0x0000 }, + { 36, 0x0008 }, + { 37, 0x000c }, + { 38, 0x0093 }, + { 39, 0x00e9 }, + { 40, 0x0000 }, + { 41, 0x0000 }, + { 42, 0x0000 }, + { 43, 0x0000 }, + { 44, 0x0003 }, + { 45, 0x0010 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0002 }, + { 50, 0x0001 }, + { 51, 0x0000 }, + { 52, 0x0000 }, + { 53, 0x0000 }, + { 54, 0x0039 }, + { 55, 0x0000 }, + { 56, 0x0001 }, }; #define WM8510_POWER1_BIASEN 0x08 @@ -58,7 +100,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { /* codec private data */ struct wm8510_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; @@ -454,6 +496,7 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3; switch (level) { @@ -467,7 +510,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8510->regmap); /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); @@ -536,10 +579,9 @@ static int wm8510_resume(struct snd_soc_codec *codec) static int wm8510_probe(struct snd_soc_codec *codec) { - struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8510->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); return ret; @@ -569,9 +611,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .suspend = wm8510_suspend, .resume = wm8510_resume, .set_bias_level = wm8510_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8510_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default =wm8510_reg, .controls = wm8510_snd_controls, .num_controls = ARRAY_SIZE(wm8510_snd_controls), @@ -586,6 +625,16 @@ static const struct of_device_id wm8510_of_match[] = { { }, }; +static const struct regmap_config wm8510_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8510_MONOMIX, + + .reg_defaults = wm8510_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8510_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -597,7 +646,10 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) if (wm8510 == NULL) return -ENOMEM; - wm8510->control_type = SND_SOC_SPI; + wm8510->regmap = devm_regmap_init_spi(spi, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + spi_set_drvdata(spi, wm8510); ret = snd_soc_register_codec(&spi->dev, @@ -635,8 +687,11 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, if (wm8510 == NULL) return -ENOMEM; + wm8510->regmap = devm_regmap_init_i2c(i2c, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + i2c_set_clientdata(i2c, wm8510); - wm8510->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/7] ASoC: wm8510: Convert to devm_kzalloc()
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 56a0495..98f28a0 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -592,7 +592,8 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&spi->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -601,8 +602,7 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } @@ -630,7 +630,8 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&i2c->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -639,8 +640,7 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } -- 1.7.10.4

1 6

[alsa-devel] [PATCH] ALSA: hda - Avoid BDL position workaround when no_period_wakeup is set
by Takashi Iwai 11 Sep '12

11 Sep '12

Originally the bogus period at BDL head was introduced as a workaround for the mismatching position update at the period boundary, typically seen on dmix. However, for applications like PulseAudio that don't require period wake ups, this workaround is just superfluous. Thus better to disable it when no_period_wakeup is given in hw_params. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6a19f6a..76ec242 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -415,6 +415,7 @@ struct azx_dev { */ unsigned int insufficient :1; unsigned int wc_marked:1; + unsigned int no_period_wakeup:1; }; /* CORB/RIRB */ @@ -1419,7 +1420,7 @@ static int azx_setup_periods(struct azx *chip, ofs = 0; azx_dev->frags = 0; pos_adj = bdl_pos_adj[chip->dev_index]; - if (pos_adj > 0) { + if (!azx_dev->no_period_wakeup && pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; int pos_align = pos_adj; pos_adj = (pos_adj * runtime->rate + 47999) / 48000; @@ -1435,8 +1436,7 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, - &bdl, ofs, pos_adj, - !substream->runtime->no_period_wakeup); + &bdl, ofs, pos_adj, true); if (ofs < 0) goto error; } @@ -1449,7 +1449,7 @@ static int azx_setup_periods(struct azx *chip, else ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes, - !substream->runtime->no_period_wakeup); + !azx_dev->no_period_wakeup); if (ofs < 0) goto error; } @@ -1922,10 +1922,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || - format_val != azx_dev->format_val) { + format_val != azx_dev->format_val || + runtime->no_period_wakeup != azx_dev->no_period_wakeup) { azx_dev->bufsize = bufsize; azx_dev->period_bytes = period_bytes; azx_dev->format_val = format_val; + azx_dev->no_period_wakeup = runtime->no_period_wakeup; err = azx_setup_periods(chip, substream, azx_dev); if (err < 0) return err; -- 1.7.11.5

1 0

[alsa-devel] [patch] ALSA: compress_core: fix open flags test in snd_compr_open()
by Dan Carpenter 11 Sep '12

11 Sep '12

O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always false and it will never do compress capture. The test for O_WRONLY is also slightly off. The original test would consider "->flags = (O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid. I've also removed the pr_err() because that could flood dmesg. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d..eb60cb8 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f) int maj = imajor(inode); int ret; - if (f->f_flags & O_WRONLY) + if ((f->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; - else if (f->f_flags & O_RDONLY) + else if ((f->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; - else { - pr_err("invalid direction\n"); + else return -EINVAL; - } if (maj == snd_major) compr = snd_lookup_minor_data(iminor(inode),

2 1

[alsa-devel] [PATCH 06/17] ARM: OMAP: Move gpio.h to include/linux/platform_data
by Tony Lindgren 11 Sep '12

11 Sep '12

This way we can remove includes of plat/gpio.h which won't work with the single zImage support. Note that we also remove the cpu_class_is_omap2() check in gpio-omap.c as the drivers should not call it as we need to make it local to arch/arm/mach-omap2 for single zImage support. While at it, arrange the related includes in the standard way. Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: linux-mtd(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- arch/arm/mach-omap1/board-ams-delta.c | 1 + arch/arm/mach-omap1/board-osk.c | 1 + arch/arm/mach-omap1/gpio15xx.c | 1 + arch/arm/mach-omap1/gpio16xx.c | 1 + arch/arm/mach-omap1/gpio7xx.c | 1 + arch/arm/mach-omap1/include/mach/gpio.h | 2 -- arch/arm/mach-omap1/leds-h2p2-debug.c | 1 + arch/arm/mach-omap1/leds.c | 1 + arch/arm/mach-omap2/board-am3517evm.c | 1 + arch/arm/mach-omap2/board-cm-t35.c | 1 + arch/arm/mach-omap2/board-zoom-display.c | 2 ++ arch/arm/mach-omap2/board-zoom-peripherals.c | 1 + arch/arm/mach-omap2/gpio.c | 1 + arch/arm/mach-omap2/hsmmc.c | 2 ++ arch/arm/mach-omap2/include/mach/gpio.h | 2 -- arch/arm/mach-omap2/msdi.c | 1 + arch/arm/mach-omap2/omap_hwmod_2420_data.c | 1 - arch/arm/mach-omap2/omap_hwmod_2430_data.c | 1 - arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c | 2 +- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 2 +- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/mach-omap2/pm24xx.c | 1 + arch/arm/mach-omap2/pm34xx.c | 2 ++ arch/arm/plat-omap/debug-leds.c | 1 + drivers/gpio/gpio-omap.c | 15 ++++++++------- drivers/mtd/nand/ams-delta.c | 8 ++++++-- include/linux/platform_data/gpio-omap.h | 15 ++------------- sound/soc/omap/sdp3430.c | 1 + 28 files changed, 40 insertions(+), 31 deletions(-) rename arch/arm/plat-omap/include/plat/gpio.h => include/linux/platform_data/gpio-omap.h (94%) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index c162369..6f192c4 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -26,6 +26,7 @@ #include <linux/export.h> #include <linux/omapfb.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <media/soc_camera.h> diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 569b687..3b2d907 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -39,6 +39,7 @@ #include <linux/mtd/partitions.h> #include <linux/mtd/physmap.h> #include <linux/i2c/tps65010.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/omap1_bl.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap1/gpio15xx.c b/arch/arm/mach-omap1/gpio15xx.c index ebef15e..98e6f39 100644 --- a/arch/arm/mach-omap1/gpio15xx.c +++ b/arch/arm/mach-omap1/gpio15xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1_MPUIO_VBASE OMAP1_MPUIO_BASE #define OMAP1510_GPIO_BASE 0xFFFCE000 diff --git a/arch/arm/mach-omap1/gpio16xx.c b/arch/arm/mach-omap1/gpio16xx.c index 2a48cd2..33f4192 100644 --- a/arch/arm/mach-omap1/gpio16xx.c +++ b/arch/arm/mach-omap1/gpio16xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1610_GPIO1_BASE 0xfffbe400 #define OMAP1610_GPIO2_BASE 0xfffbec00 diff --git a/arch/arm/mach-omap1/gpio7xx.c b/arch/arm/mach-omap1/gpio7xx.c index acf12b7..958ce9a 100644 --- a/arch/arm/mach-omap1/gpio7xx.c +++ b/arch/arm/mach-omap1/gpio7xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP7XX_GPIO1_BASE 0xfffbc000 #define OMAP7XX_GPIO2_BASE 0xfffbc800 diff --git a/arch/arm/mach-omap1/include/mach/gpio.h b/arch/arm/mach-omap1/include/mach/gpio.h index e737706..ebf86c0 100644 --- a/arch/arm/mach-omap1/include/mach/gpio.h +++ b/arch/arm/mach-omap1/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap1/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap1/leds-h2p2-debug.c b/arch/arm/mach-omap1/leds-h2p2-debug.c index f6b14a1..7f4bba9 100644 --- a/arch/arm/mach-omap1/leds-h2p2-debug.c +++ b/arch/arm/mach-omap1/leds-h2p2-debug.c @@ -14,6 +14,7 @@ #include <linux/kernel_stat.h> #include <linux/sched.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <mach/hardware.h> #include <asm/leds.h> diff --git a/arch/arm/mach-omap1/leds.c b/arch/arm/mach-omap1/leds.c index ae6dd93..7b1a383 100644 --- a/arch/arm/mach-omap1/leds.c +++ b/arch/arm/mach-omap1/leds.c @@ -6,6 +6,7 @@ #include <linux/gpio.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/leds.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap2/board-am3517evm.c b/arch/arm/mach-omap2/board-am3517evm.c index 00abda1..029ebdf 100644 --- a/arch/arm/mach-omap2/board-am3517evm.c +++ b/arch/arm/mach-omap2/board-am3517evm.c @@ -25,6 +25,7 @@ #include <linux/can/platform/ti_hecc.h> #include <linux/davinci_emac.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> #include <mach/hardware.h> #include <mach/am35xx.h> diff --git a/arch/arm/mach-omap2/board-cm-t35.c b/arch/arm/mach-omap2/board-cm-t35.c index ea34109..34cb904 100644 --- a/arch/arm/mach-omap2/board-cm-t35.c +++ b/arch/arm/mach-omap2/board-cm-t35.c @@ -23,6 +23,7 @@ #include <linux/input/matrix_keypad.h> #include <linux/delay.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/i2c/at24.h> #include <linux/i2c/twl.h> diff --git a/arch/arm/mach-omap2/board-zoom-display.c b/arch/arm/mach-omap2/board-zoom-display.c index 28187f1..d64f35f 100644 --- a/arch/arm/mach-omap2/board-zoom-display.c +++ b/arch/arm/mach-omap2/board-zoom-display.c @@ -18,6 +18,8 @@ #include <video/omapdss.h> #include <mach/board-zoom.h> +#include <plat/cpu.h> + #define LCD_PANEL_RESET_GPIO_PROD 96 #define LCD_PANEL_RESET_GPIO_PILOT 55 #define LCD_PANEL_QVGA_GPIO 56 diff --git a/arch/arm/mach-omap2/board-zoom-peripherals.c b/arch/arm/mach-omap2/board-zoom-peripherals.c index 00f73b7..ae5d36b 100644 --- a/arch/arm/mach-omap2/board-zoom-peripherals.c +++ b/arch/arm/mach-omap2/board-zoom-peripherals.c @@ -19,6 +19,7 @@ #include <linux/regulator/fixed.h> #include <linux/wl12xx.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> diff --git a/arch/arm/mach-omap2/gpio.c b/arch/arm/mach-omap2/gpio.c index 9ad7d48..7b52bdf 100644 --- a/arch/arm/mach-omap2/gpio.c +++ b/arch/arm/mach-omap2/gpio.c @@ -21,6 +21,7 @@ #include <linux/slab.h> #include <linux/interrupt.h> #include <linux/of.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/hsmmc.c b/arch/arm/mach-omap2/hsmmc.c index a9675d8..80399d7 100644 --- a/arch/arm/mach-omap2/hsmmc.c +++ b/arch/arm/mach-omap2/hsmmc.c @@ -15,6 +15,8 @@ #include <linux/delay.h> #include <linux/gpio.h> #include <mach/hardware.h> +#include <linux/platform_data/gpio-omap.h> + #include <plat/mmc.h> #include <plat/omap-pm.h> #include <plat/mux.h> diff --git a/arch/arm/mach-omap2/include/mach/gpio.h b/arch/arm/mach-omap2/include/mach/gpio.h index be4d290..5621cc5 100644 --- a/arch/arm/mach-omap2/include/mach/gpio.h +++ b/arch/arm/mach-omap2/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap2/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap2/msdi.c b/arch/arm/mach-omap2/msdi.c index fb5bc6c..9e57b4a 100644 --- a/arch/arm/mach-omap2/msdi.c +++ b/arch/arm/mach-omap2/msdi.c @@ -23,6 +23,7 @@ #include <linux/kernel.h> #include <linux/err.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2420_data.c b/arch/arm/mach-omap2/omap_hwmod_2420_data.c index 50cfab6..5b49ebf 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2420_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2420_data.c @@ -18,7 +18,6 @@ #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mcspi.h> #include <plat/dmtimer.h> #include <plat/l3_2xxx.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2430_data.c b/arch/arm/mach-omap2/omap_hwmod_2430_data.c index 58b5bc1..1fd66f0 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2430_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2430_data.c @@ -18,7 +18,6 @@ #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mcbsp.h> #include <plat/mcspi.h> #include <plat/dmtimer.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c index afad69c..c6416c9 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c @@ -10,7 +10,7 @@ */ #include <plat/omap_hwmod.h> #include <plat/serial.h> -#include <plat/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/dma.h> #include <plat/dmtimer.h> #include <plat/mcspi.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c index c9e3820..5820c73 100644 --- a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c @@ -15,6 +15,7 @@ * XXX these should be marked initdata for multi-OMAP kernels */ #include <linux/power/smartreflex.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <mach/irqs.h> @@ -24,7 +25,6 @@ #include <plat/l3_3xxx.h> #include <plat/l4_3xxx.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mmc.h> #include <plat/mcbsp.h> #include <plat/mcspi.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c index 242aee4..fe987f8 100644 --- a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c @@ -19,12 +19,12 @@ */ #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/power/smartreflex.h> #include <plat/omap_hwmod.h> #include <plat/cpu.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/dma.h> #include <plat/mcspi.h> #include <plat/mcbsp.h> diff --git a/arch/arm/mach-omap2/pm24xx.c b/arch/arm/mach-omap2/pm24xx.c index de1caf1..2708a8b 100644 --- a/arch/arm/mach-omap2/pm24xx.c +++ b/arch/arm/mach-omap2/pm24xx.c @@ -29,6 +29,7 @@ #include <linux/irq.h> #include <linux/time.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach/time.h> #include <asm/mach/irq.h> diff --git a/arch/arm/mach-omap2/pm34xx.c b/arch/arm/mach-omap2/pm34xx.c index 05bd8f0..90480f7 100644 --- a/arch/arm/mach-omap2/pm34xx.c +++ b/arch/arm/mach-omap2/pm34xx.c @@ -28,6 +28,8 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/platform_data/gpio-omap.h> + #include <trace/events/power.h> #include <asm/suspend.h> diff --git a/arch/arm/plat-omap/debug-leds.c b/arch/arm/plat-omap/debug-leds.c index 39407cb..195aaae 100644 --- a/arch/arm/plat-omap/debug-leds.c +++ b/arch/arm/plat-omap/debug-leds.c @@ -12,6 +12,7 @@ #include <linux/platform_device.h> #include <linux/leds.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <mach/hardware.h> #include <asm/leds.h> diff --git a/drivers/gpio/gpio-omap.c b/drivers/gpio/gpio-omap.c index e6efd77..64fbce3 100644 --- a/drivers/gpio/gpio-omap.c +++ b/drivers/gpio/gpio-omap.c @@ -25,11 +25,9 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/irqdomain.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> -#include <asm/irq.h> -#include <mach/irqs.h> -#include <asm/gpio.h> #include <asm/mach/irq.h> #define OFF_MODE 1 @@ -385,13 +383,16 @@ static int _set_gpio_triggering(struct gpio_bank *bank, int gpio, static int gpio_irq_type(struct irq_data *d, unsigned type) { struct gpio_bank *bank = irq_data_get_irq_chip_data(d); - unsigned gpio; + unsigned gpio = 0; int retval; unsigned long flags; - if (!cpu_class_is_omap2() && d->irq > IH_MPUIO_BASE) +#ifdef CONFIG_ARCH_OMAP1 + if (d->irq > IH_MPUIO_BASE) gpio = OMAP_MPUIO(d->irq - IH_MPUIO_BASE); - else +#endif + + if (!gpio) gpio = irq_to_gpio(bank, d->irq); if (type & ~IRQ_TYPE_SENSE_MASK) diff --git a/drivers/mtd/nand/ams-delta.c b/drivers/mtd/nand/ams-delta.c index 861ca8f..78a524b 100644 --- a/drivers/mtd/nand/ams-delta.c +++ b/drivers/mtd/nand/ams-delta.c @@ -23,12 +23,16 @@ #include <linux/mtd/mtd.h> #include <linux/mtd/nand.h> #include <linux/mtd/partitions.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> + #include <asm/io.h> -#include <mach/hardware.h> #include <asm/sizes.h> -#include <linux/gpio.h> + #include <plat/board-ams-delta.h> +#include <mach/hardware.h> + /* * MTD structure for E3 (Delta) */ diff --git a/arch/arm/plat-omap/include/plat/gpio.h b/include/linux/platform_data/gpio-omap.h similarity index 94% rename from arch/arm/plat-omap/include/plat/gpio.h rename to include/linux/platform_data/gpio-omap.h index 50fb7cc..e8741c2 100644 --- a/arch/arm/plat-omap/include/plat/gpio.h +++ b/include/linux/platform_data/gpio-omap.h @@ -1,6 +1,4 @@ /* - * arch/arm/plat-omap/include/mach/gpio.h - * * OMAP GPIO handling defines and functions * * Copyright (C) 2003-2005 Nokia Corporation @@ -155,6 +153,8 @@ #define OMAP4_GPIO_CLEARDATAOUT 0x0190 #define OMAP4_GPIO_SETDATAOUT 0x0194 +#define OMAP_MAX_GPIO_LINES 192 + #define OMAP_MPUIO(nr) (OMAP_MAX_GPIO_LINES + (nr)) #define OMAP_GPIO_IS_MPUIO(nr) ((nr) >= OMAP_MAX_GPIO_LINES) @@ -213,16 +213,5 @@ extern void omap2_gpio_prepare_for_idle(int off_mode); extern void omap2_gpio_resume_after_idle(void); extern void omap_set_gpio_debounce(int gpio, int enable); extern void omap_set_gpio_debounce_time(int gpio, int enable); -/*-------------------------------------------------------------------------*/ - -/* - * Wrappers for "new style" GPIO calls, using the new infrastructure - * which lets us plug in FPGA, I2C, and other implementations. - * - * The original OMAP-specific calls should eventually be removed. - */ - -#include <linux/errno.h> -#include <asm-generic/gpio.h> #endif diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 0e28322..78e1419 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -34,6 +34,7 @@ #include <mach/hardware.h> #include <mach/gpio.h> #include <plat/mcbsp.h> +#include <linux/platform_data/gpio-omap.h> /* Register descriptions for twl4030 codec part */ #include <linux/mfd/twl4030-audio.h>

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[alsa-devel] Need help with failed driver initialization
by Timur Tabi 11 Sep '12

11 Sep '12

Every 6-12 months it seems, my audio drivers break because of some update to the ASoC. It's that time again. It appears that support for deferred probing isn't working for me. I turned on debug printks (and added a few of my own to my drivers), and here's the log: It looks like everything is registering, but the audio device still doesn't exist. Can anyone give me a clue what's wrong? bus: 'i2c': driver_probe_device: matched device 0-004f with driver cs4270 bus: 'i2c': really_probe: probing driver cs4270 with device 0-004f cs4270 0-004f: found device at i2c address 4F cs4270 0-004f: hardware revision 3 cs4270 0-004f: codec register 0-004f cs4270 0-004f: dai register 0-004f #1 cs4270 0-004f: Mapped DAI cs4270-hifi to CODEC cs4270.0-004f Registered DAI 'cs4270-hifi' Registered codec 'cs4270.0-004f' driver: '0-004f': driver_bound: bound to device 'cs4270' bus: 'i2c': really_probe: bound device 0-004f to driver cs4270 Freescale MPC8610 HPCD ALSA SoC machine driver bus: 'platform': driver_probe_device: matched device e0016000.ssi with driver fsl-ssi-dai bus: 'platform': really_probe: probing driver fsl-ssi-dai with device e0016000.ssi fsl-ssi-dai e0016000.ssi: probing fsl-ssi-dai e0016000.ssi: registering with ASoC fsl-ssi-dai e0016000.ssi: dai register e0016000.ssi Registered DAI 'e0016000.ssi' fsl-ssi-dai e0016000.ssi: AsoC registration complete fsl-ssi-dai e0016000.ssi: platform registration 'snd-soc-mpc8610hpcd' bus: 'platform': driver_probe_device: matched device snd-soc-mpc8610hpcd.0 with driver snd-soc-mpc8610hpcd bus: 'platform': really_probe: probing driver snd-soc-mpc8610hpcd with device snd-soc-mpc8610hpcd.0 bus: 'platform': driver_probe_device: matched device soc-audio with driver soc-audio bus: 'platform': really_probe: probing driver soc-audio with device soc-audio soc-audio soc-audio: ASoC machine snd-soc-mpc8610hpcd should use snd_soc_register_card() soc-audio soc-audio: binding playback at idx 0 soc-audio soc-audio: platform e0021100.dma-channel not registered platform soc-audio: Driver soc-audio requests probe deferral platform soc-audio: Added to deferred list driver: 'snd-soc-mpc8610hpcd.0': driver_bound: bound to device 'snd-soc-mpc8610hpcd' bus: 'platform': really_probe: bound device snd-soc-mpc8610hpcd.0 to driver snd-soc-mpc8610hpcd fsl-ssi-dai e0016000.ssi: probe successful driver: 'e0016000.ssi': driver_bound: bound to device 'fsl-ssi-dai' bus: 'platform': really_probe: bound device e0016000.ssi to driver fsl-ssi-dai bus: 'platform': driver_probe_device: matched device e0021100.dma-channel with driver fsl-pcm-audio bus: 'platform': really_probe: probing driver fsl-pcm-audio with device e0021100.dma-channel fsl-pcm-audio e0021100.dma-channel: probing fsl-pcm-audio e0021100.dma-channel: platform register e0021100.dma-channel Registered platform 'e0021100.dma-channel' fsl-pcm-audio e0021100.dma-channel: probe successful driver: 'e0021100.dma-channel': driver_bound: bound to device 'fsl-pcm-audio' bus: 'platform': really_probe: bound device e0021100.dma-channel to driver fsl-pcm-audio bus: 'platform': driver_probe_device: matched device e0021180.dma-channel with driver fsl-pcm-audio bus: 'platform': really_probe: probing driver fsl-pcm-audio with device e0021180.dma-channel fsl-pcm-audio e0021180.dma-channel: probing fsl-pcm-audio e0021180.dma-channel: platform register e0021180.dma-channel Registered platform 'e0021180.dma-channel' fsl-pcm-audio e0021180.dma-channel: probe successful driver: 'e0021180.dma-channel': driver_bound: bound to device 'fsl-pcm-audio' bus: 'platform': really_probe: bound device e0021180.dma-channel to driver fsl-pcm-audio ... ALSA device list: No soundcards found. platform soc-audio: Retrying from deferred list bus: 'platform': driver_probe_device: matched device soc-audio with driver soc-audio bus: 'platform': really_probe: probing driver soc-audio with device soc-audio soc-audio: probe of soc-audio failed with error -22 When I use speaker-test, I get this: # speaker-test -t sine -c 2 -r 48000 speaker-test 1.0.11rc2 Playback device is default Stream parameters are 48000Hz, S16_BE, 2 channels Sine wave rate is 440.0000Hz ALSA lib confmisc.c:768:(parse_card) cannot find card '0' ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2202:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [asoc:for-next 110/115] sound/soc/soc-core.c:901 soc_bind_dai_link() error: we previously assumed 'dai_link->platform_of_node' could be null (see line 887)
by Fengguang Wu 11 Sep '12

11 Sep '12

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: bffb28a4233d25d37a444785f766e98a041b4413 commit: a6179d2741640a1ec88a1c5b9c2c28bdf45666db [110/115] ASoC: core: display the platform node name if there is no platform name All smatch warnings: + sound/soc/soc-core.c:901 soc_bind_dai_link() error: we previously assumed 'dai_link->platform_of_node' could be null (see line 887) vim +901 sound/soc/soc-core.c 884 885 /* find one from the set of registered platforms */ 886 list_for_each_entry(platform, &platform_list, list) { > 887 if (dai_link->platform_of_node) { 888 if (platform->dev->of_node != 889 dai_link->platform_of_node) 890 continue; 891 } else { 892 if (strcmp(platform->name, platform_name)) 893 continue; 894 } 895 896 rtd->platform = platform; 897 } 898 if (!rtd->platform) { 899 dev_err(card->dev, "platform %s not registered\n", 900 dai_link->platform_name ? : > 901 dai_link->platform_of_node->full_name); 902 return -EPROBE_DEFER; 903 } 904 905 card->num_rtd++; 906 907 return 0; 908 } --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

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[alsa-devel] [PATCH 1/3] ASoC: wm8900: Convert to devm_kzalloc()
by Mark Brown 11 Sep '12

11 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8900.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 077c962..5f7a78e 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1226,7 +1226,8 @@ static int __devinit wm8900_spi_probe(struct spi_device *spi) struct wm8900_priv *wm8900; int ret; - wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + wm8900 = devm_kzalloc(&spi->dev, sizeof(struct wm8900_priv), + GFP_KERNEL); if (wm8900 == NULL) return -ENOMEM; @@ -1235,15 +1236,13 @@ static int __devinit wm8900_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8900, &wm8900_dai, 1); - if (ret < 0) - kfree(wm8900); + return ret; } static int __devexit wm8900_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1264,7 +1263,8 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, struct wm8900_priv *wm8900; int ret; - wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + wm8900 = devm_kzalloc(&i2c->dev, sizeof(struct wm8900_priv), + GFP_KERNEL); if (wm8900 == NULL) return -ENOMEM; @@ -1273,15 +1273,13 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8900, &wm8900_dai, 1); - if (ret < 0) - kfree(wm8900); + return ret; } static __devexit int wm8900_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: core: display the platform node name if there is no platform name
by Timur Tabi 11 Sep '12

11 Sep '12

A platform can be matched by either a string name or a device tree node pointer, so we can't assume that dai_link->platform_name is valid. Fix an error message in soc_bind_dai_link() to use the device tree node name if the platform name is null. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad65459..d81ef5b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -897,7 +897,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->platform) { dev_err(card->dev, "platform %s not registered\n", - dai_link->platform_name); + dai_link->platform_name ? : + dai_link->platform_of_node->full_name); return -EPROBE_DEFER; } -- 1.7.3.4

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[alsa-devel] [PATCH 1/3] ASoC: Revert "ASoC: mc13783: Provide codec->control_data"
by Fabio Estevam 11 Sep '12

11 Sep '12

Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap) , it is not necessary to provide codec->control_data anymore. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/mc13783.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8def29a..bc95599 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -585,8 +585,6 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - codec->control_data = priv->mc13xxx; - mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: am3517evm: Remove unused cpu_dai from hw_params
by Peter Ujfalusi 11 Sep '12

11 Sep '12

cpu_dai is not in use in this function and just generates warning at compile time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Commit: fca04ae ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing left this line in the code which generates now warning. This patch is on top of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap Regards, Peter sound/soc/omap/am3517evm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index a997988..77a8701 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -41,7 +41,6 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; /* Set the codec system clock for DAC and ADC */ -- 1.7.12

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[alsa-devel] [PATCH] ASoC: tegra: fix maxburst settings in dmaengine code
by Stephen Warren 11 Sep '12

11 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> The I2S controllers are programmed with an "attention" level of 4 DWORDs. This must match the configuration passed to the DMA driver, so that when they burst in data, they don't overflow the available FIFO space. Also, the burst size is relevant to the destination for playback, and source for capture, not vice-versa as originally written. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, I'd like to take this patch through the Tegra tree, since it's a dependency for the series that converts Tegra to dmaengine. Thanks. sound/soc/tegra/tegra_pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 5658bce..8d6900c 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.dst_addr = dmap->addr; - slave_config.src_maxburst = 0; + slave_config.dst_maxburst = 4; } else { slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.src_addr = dmap->addr; - slave_config.dst_maxburst = 0; + slave_config.src_maxburst = 4; } slave_config.slave_id = dmap->req_sel; -- 1.7.0.4

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[alsa-devel] snd-intel-hda: screaming IRQ with 3.5.3 kernel (HDMI)
by Chris Rankin 11 Sep '12

11 Sep '12

Hi, I have recently added a Radeon HD4670 to one of my older PCs, and have noticed that it has IRQ trouble when I try to enable the HDMI audio: irq 17: nobody cared (try booting with the "irqpoll" option) Pid: 0, comm: swapper/0 Not tainted 3.5.3 #2 Call Trace: [<c1063806>] ? __report_bad_irq+0x11/0x94 [<c1063a2d>] ? note_interrupt+0x120/0x19b [<c1062325>] ? handle_irq_event_percpu+0xe8/0xfb [<c1062361>] ? handle_irq_event+0x29/0x40 [<c10640fc>] ? handle_level_irq+0x93/0x93 [<c1064163>] ? handle_fasteoi_irq+0x67/0x90 <IRQ> [<c1003625>] ? do_IRQ+0x2e/0x83 [<c1229829>] ? common_interrupt+0x29/0x30 [<c106007b>] ? audit_update_watch+0x59/0x285 [<c1007479>] ? default_idle+0x23/0x3b [<c1007b38>] ? cpu_idle+0x4b/0x7e [<c1314877>] ? start_kernel+0x2a4/0x2a7 handlers: [<f8c97b75>] azx_interrupt [snd_hda_intel] Disabling IRQ #17 This is frequently followed by KMS trouble too: irq 16: nobody cared (try booting with the "irqpoll" option) Pid: 0, comm: swapper/0 Not tainted 3.5.3 #2 Call Trace: [<c1063806>] ? __report_bad_irq+0x11/0x94 [<c1063a2d>] ? note_interrupt+0x120/0x19b [<c1062325>] ? handle_irq_event_percpu+0xe8/0xfb [<c1062361>] ? handle_irq_event+0x29/0x40 [<c10640fc>] ? handle_level_irq+0x93/0x93 [<c1064163>] ? handle_fasteoi_irq+0x67/0x90 <IRQ> [<c1003625>] ? do_IRQ+0x2e/0x83 [<c1229829>] ? common_interrupt+0x29/0x30 [<c106007b>] ? audit_update_watch+0x59/0x285 [<c1007479>] ? default_idle+0x23/0x3b [<c1007b38>] ? cpu_idle+0x4b/0x7e [<c1314877>] ? start_kernel+0x2a4/0x2a7 handlers: [<f818bfb3>] radeon_driver_irq_handler_kms [radeon] [<f81fe55f>] usb_hcd_irq [usbcore] Disabling IRQ #16 The PCI entries are: 01:00.1 Audio device: ATI Technologies Inc RV710/730 Subsystem: ATI Technologies Inc RV710/730 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (2000ns min), Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 17 Region 0: Memory at ff8ec000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: snd_hda_intel and: 01:00.0 VGA compatible controller: ATI Technologies Inc RV730 Pro AGP [Radeon HD 4600 Series] (prog-if 00 [VGA controller]) Subsystem: ATI Technologies Inc Device 0028 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx+ Latency: 64 (2000ns min), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at e0000000 (32-bit, prefetchable) [size=256M] Region 1: I/O ports at ec00 [size=256] Region 2: Memory at ff8f0000 (32-bit, non-prefetchable) [size=64K] Expansion ROM at ff800000 [disabled] [size=128K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] AGP version 3.0 Status: RQ=256 Iso- ArqSz=0 Cal=0 SBA+ ITACoh- GART64- HTrans- 64bit- FW+ AGP3+ Rate=x4,x8 Command: RQ=32 ArqSz=2 Cal=0 SBA+ AGP+ GART64- 64bit- FW- Rate=x8 Kernel driver in use: radeon Does anyone have any suggestions, please? Thanks, Chris

2 5

[alsa-devel] [PATCH 1/3] ASoC: cs4270: Conver to data based control init
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 44a176f..c6e5a73 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -521,14 +521,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) return ret; } - /* Add the non-DAPM controls */ - ret = snd_soc_add_codec_controls(codec, cs4270_snd_controls, - ARRAY_SIZE(cs4270_snd_controls)); - if (ret < 0) { - dev_err(codec->dev, "failed to add controls\n"); - return ret; - } - /* get the power supply regulators */ for (i = 0; i < ARRAY_SIZE(supply_names); i++) cs4270->supplies[i].supply = supply_names[i]; @@ -634,6 +626,9 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .remove = cs4270_remove, .suspend = cs4270_soc_suspend, .resume = cs4270_soc_resume, + + .controls = cs4270_snd_controls, + .num_controls = ARRAY_SIZE(cs4270_snd_controls), .volatile_register = cs4270_reg_is_volatile, .readable_register = cs4270_reg_is_readable, .reg_cache_size = CS4270_LASTREG + 1, -- 1.7.10.4

2 4

Re: [alsa-devel] 3.6-rc Cirrus HDA reconfig oops...
by Takashi Iwai 10 Sep '12

10 Sep '12

At Mon, 10 Sep 2012 23:00:39 +0800, Daniel J Blueman wrote: > > On 10 September 2012 16:29, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 10 Sep 2012 09:46:43 +0200, > > Takashi Iwai wrote: > >> > >> At Sun, 9 Sep 2012 13:02:16 +0800, > >> Daniel J Blueman wrote: > >> > > >> > With a non-specialised Cirrus HDA codec, it's readily possible to > >> > cause a kernel oops as root with 3.6-rc kernels [1]. > >> > > >> > patch_cirrus.c: > >> > > >> > static int cs_init(struct hda_codec *codec) > >> > { > >> > struct cs_spec *spec = codec->spec; > >> > > >> > /* init_verb sequence for C0/C1/C2 errata*/ > >> > snd_hda_sequence_write(codec, cs_errata_init_verbs); > >> > > >> > snd_hda_sequence_write(codec, cs_coef_init_verbs); > >> > > >> > if (spec->gpio_mask) { > >> > snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, > >> > spec->gpio_mask); > >> > > >> > Here, spec is NULL, so loading spec->gpio_mask causes a fatal > >> > pagefault at address 0x180 (ie the offset of gpio_mask in the struct). > >> > I was going to prepare a patch to guard for this, but since spec is > >> > expected in so many places, there is a likely a behavioural issue > >> > preventing a spec struct being generated. > >> > > >> > Let me know for further testing and debug. > >> > >> It's because of calling a leftover callback. > >> The patch below should fix the issue. I'll queue it up to for-linus > >> branch. > > > > Oops, a wrong patch. The corrected one below. > > [...] > > Superb; this fixes the issue nicely. > > Thanks Takashi! OK, thanks for confirmation! Takashi

1 0

Re: [alsa-devel] UAC2 clock selector for AudioBox 1818VSL
by Daniel Mack 10 Sep '12

10 Sep '12

On 07.09.2012 19:57, louis(a)museresearch.com wrote: > Hi Daniel, > Thanks for the quick reply. I am able to set the clock on my Mac, though > I do need to do further tests and make sure that the sync is actually > reliable there. The setting is available from the built-in, standard OS > X audio setup application "Audio MIDI Setup". > > There should be a S/PDIF clock signal going to the digital inputs on the > device. Some of the other controls such as per-channel mute also appear to > work. I've also attached the lsusb output. > > There are some other details that may be relevant, but I didn't want to > make my initial post too long/confusing because these extra details may > end up just being a red herring: Each mixer element name (as seen in > amixer) is prefixed with "AudioBox 1818 VSL Clock Selector" even if it is > not a clock selector. That strange but I believe not related. We should still fix that :) > I am confident though that the control I've been > playing with is the clock selector because it lists the different clock > types as options. I also see this when I plug the device in or load the > ALSA driver: > > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x201, wIndex = 0xa00, type = 4 > ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 > (id 10) > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x200, wIndex = 0xa00, type = 4 > ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 > (id 10) > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x201, wIndex = 0xb00, type = 4 > ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 > (id 11) > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x200, wIndex = 0xb00, type = 4 > ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 > (id 11) Do you see any messages when you try to set the clock selector with amixer? If you're running a recent kernel, try the following patch. That will produce log messages for each mixer related control packet that is sent out to the device. Thanks, Daniel

2 1

[alsa-devel] Invitation to connect on LinkedIn
by harsha priya gupta via LinkedIn 10 Sep '12

10 Sep '12

LinkedIn ------------ harsha priya gupta requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - harsha priya Accept invitation from harsha priya gupta http://www.linkedin.com/e/jpxkk3-h6xtnnj8-6b/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of harsha priya gupta http://www.linkedin.com/e/jpxkk3-h6xtnnj8-6b/rso/147246195/LJfd/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-h6xtnnj8-6b/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH 05/22] ASoC: Ux500: Enable MOP500 driver for Device Tree
by Lee Jones 10 Sep '12

10 Sep '12

Here we ensure that the MOP500 audio driver will be probed during a Device Tree boot. We also parse the sound node to link together the codec, dma and the CPU-side Digital Audio Interface. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/mop500.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 31c4d26..6840df7 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -16,6 +16,7 @@ #include <linux/module.h> #include <linux/io.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <sound/soc.h> #include <sound/initval.h> @@ -56,8 +57,35 @@ static struct snd_soc_card mop500_card = { .num_links = ARRAY_SIZE(mop500_dai_links), }; +static int __devinit mop500_of_probe(struct platform_device *pdev, + struct device_node *np) +{ + struct device_node *codec_np, *msp_np[2]; + int i; + + msp_np[0] = of_parse_phandle(np, "stericsson,cpu-dai", 0); + msp_np[1] = of_parse_phandle(np, "stericsson,cpu-dai", 1); + codec_np = of_parse_phandle(np, "stericsson,audio-codec", 0); + + if (!(msp_np[0] && msp_np[1] && codec_np)) { + dev_err(&pdev->dev, "Phandle missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < 2; i++) { + mop500_dai_links[i].cpu_of_node = msp_np[i]; + mop500_dai_links[i].cpu_dai_name = NULL; + mop500_dai_links[i].codec_of_node = codec_np; + mop500_dai_links[i].codec_name = NULL; + } + + snd_soc_of_parse_card_name(&mop500_card, "stericsson,card-name"); + + return 0; +} static int __devinit mop500_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; int ret; pr_debug("%s: Enter.\n", __func__); @@ -66,6 +94,12 @@ static int __devinit mop500_probe(struct platform_device *pdev) mop500_card.dev = &pdev->dev; + if (np) { + ret = mop500_of_probe(pdev, np); + if (ret) + return ret; + } + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", __func__, mop500_card.name); platform_set_drvdata(pdev, &mop500_card); @@ -101,10 +135,16 @@ static int __devexit mop500_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id snd_soc_mop500_match[] = { + { .compatible = "stericsson,snd-soc-mop500", }, + {}, +}; + static struct platform_driver snd_soc_mop500_driver = { .driver = { .owner = THIS_MODULE, .name = "snd-soc-mop500", + .of_match_table = snd_soc_mop500_match, }, .probe = mop500_probe, .remove = __devexit_p(mop500_remove), -- 1.7.9.5

2 2

[alsa-devel] [PATCH 06/22] ASoC: Ux500: Enable ux500 MSP driver for Device Tree
by Lee Jones 10 Sep '12

10 Sep '12

Register both parts of the MSP driver from Device Tree so that they are probed when Device Tree is enabled. Also, as there is platform data involved, we ensure that there is allocated memory to place the configuration into and that the correct information is extracted from the DT binary. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_msp_dai.c | 6 ++++++ sound/soc/ux500/ux500_msp_i2s.c | 22 +++++++++++++++++++--- 2 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 772cb19..0f7dd49 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -833,10 +833,16 @@ static int __devexit ux500_msp_drv_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id ux500_msp_i2c_match[] = { + { .compatible = "stericsson,ux500-msp-i2s", }, + {}, +}; + static struct platform_driver msp_i2s_driver = { .driver = { .name = "ux500-msp-i2s", .owner = THIS_MODULE, + .of_match_table = ux500_msp_i2c_match, }, .probe = ux500_msp_drv_probe, .remove = ux500_msp_drv_remove, diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 2cbfc54..5e0bf8c 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -18,6 +18,7 @@ #include <linux/pinctrl/consumer.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of.h> #include <mach/hardware.h> #include <mach/msp.h> @@ -692,16 +693,31 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, int ret = 0; struct resource *res = NULL; struct i2s_controller *i2s_cont; + struct device_node *np = pdev->dev.of_node; struct ux500_msp *msp; - dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, - pdev->name, platform_data->id); - *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); msp = *msp_p; if (!msp) return -ENOMEM; + if (np) { + if (!platform_data) { + platform_data = devm_kzalloc(&pdev->dev, + sizeof(struct msp_i2s_platform_data), GFP_KERNEL); + if (!platform_data) + ret = -ENOMEM; + } + } else + if (!platform_data) + ret = -EINVAL; + + if (ret) + goto err_res; + + dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, + pdev->name, platform_data->id); + msp->id = platform_data->id; msp->dev = &pdev->dev; msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: wm8960: Convert to direct regmap API usage
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8960.c | 107 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 84 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 804f411..7cb0d07c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -52,25 +52,72 @@ * We can't read the WM8960 register space when we are * using 2 wire for device control, so we cache them instead. */ -static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0000, 0x0000, - 0x0000, 0x0008, 0x0000, 0x000a, - 0x01c0, 0x0000, 0x00ff, 0x00ff, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x007b, 0x0100, 0x0032, - 0x0000, 0x00c3, 0x00c3, 0x01c0, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0100, 0x0100, 0x0050, 0x0050, - 0x0050, 0x0050, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0040, 0x0000, - 0x0000, 0x0050, 0x0050, 0x0000, - 0x0002, 0x0037, 0x004d, 0x0080, - 0x0008, 0x0031, 0x0026, 0x00e9, +static const struct reg_default wm8960_reg_defaults[] = { + { 0x0, 0x0097 }, + { 0x1, 0x0097 }, + { 0x2, 0x0000 }, + { 0x3, 0x0000 }, + { 0x4, 0x0000 }, + { 0x5, 0x0008 }, + { 0x6, 0x0000 }, + { 0x7, 0x000a }, + { 0x8, 0x01c0 }, + { 0x9, 0x0000 }, + { 0xa, 0x00ff }, + { 0xb, 0x00ff }, + + { 0x10, 0x0000 }, + { 0x11, 0x007b }, + { 0x12, 0x0100 }, + { 0x13, 0x0032 }, + { 0x14, 0x0000 }, + { 0x15, 0x00c3 }, + { 0x16, 0x00c3 }, + { 0x17, 0x01c0 }, + { 0x18, 0x0000 }, + { 0x19, 0x0000 }, + { 0x1a, 0x0000 }, + { 0x1b, 0x0000 }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + + { 0x20, 0x0100 }, + { 0x21, 0x0100 }, + { 0x22, 0x0050 }, + + { 0x25, 0x0050 }, + { 0x26, 0x0000 }, + { 0x27, 0x0000 }, + { 0x28, 0x0000 }, + { 0x29, 0x0000 }, + { 0x2a, 0x0040 }, + { 0x2b, 0x0000 }, + { 0x2c, 0x0000 }, + { 0x2d, 0x0050 }, + { 0x2e, 0x0050 }, + { 0x2f, 0x0000 }, + { 0x30, 0x0002 }, + { 0x31, 0x0037 }, + + { 0x33, 0x0080 }, + { 0x34, 0x0008 }, + { 0x35, 0x0031 }, + { 0x36, 0x0026 }, + { 0x37, 0x00e9 }, }; +static bool wm8960_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8960_RESET: + return true; + default: + return false; + } +} + struct wm8960_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); struct snd_soc_dapm_widget *lout1; @@ -555,6 +602,8 @@ static int wm8960_mute(struct snd_soc_dai *dai, int mute) static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -566,7 +615,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8960->regmap); /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, @@ -667,7 +716,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - snd_soc_cache_sync(codec); + regcache_sync(wm8960->regmap); break; default: break; @@ -915,7 +964,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level = wm8960_set_bias_level_capless; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8960->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -963,9 +1012,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8960 = { .suspend = wm8960_suspend, .resume = wm8960_resume, .set_bias_level = wm8960_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8960_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8960_reg, +}; + +static const struct regmap_config wm8960_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8960_PLL4, + + .reg_defaults = wm8960_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8960_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8960_volatile, }; static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, @@ -979,8 +1037,11 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->regmap = regmap_init_i2c(i2c, &wm8960_regmap); + if (IS_ERR(wm8960->regmap)) + return PTR_ERR(wm8960->regmap); + i2c_set_clientdata(i2c, wm8960); - wm8960->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8904: remove redundant code
by Bo Shen 10 Sep '12

10 Sep '12

The core_intercon is added two times, remove the redundant one Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 0013afe..2b9766d 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1185,8 +1185,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets, ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, core_intercon, - ARRAY_SIZE(core_intercon)); snd_soc_dapm_add_routes(dapm, adc_intercon, ARRAY_SIZE(adc_intercon)); snd_soc_dapm_add_routes(dapm, dac_intercon, -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8741: Move regulator acquisition to I2C/SPI probe()
by Mark Brown 10 Sep '12

10 Sep '12

Better style as we acquire resources before trying the ASoC card probe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8741.c | 37 ++++++++++++++++++++++--------------- 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 35f3d23..742744b 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -403,17 +403,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); int ret = 0; - int i; - - for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) - wm8741->supplies[i].supply = wm8741_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; - } ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); @@ -450,7 +439,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err: return ret; } @@ -460,7 +448,6 @@ static int wm8741_remove(struct snd_soc_codec *codec) struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); return 0; } @@ -492,13 +479,23 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8741_priv *wm8741; - int ret; + int ret, i; wm8741 = devm_kzalloc(&i2c->dev, sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + i2c_set_clientdata(i2c, wm8741); wm8741->control_type = SND_SOC_I2C; @@ -536,13 +533,23 @@ static struct i2c_driver wm8741_i2c_driver = { static int __devinit wm8741_spi_probe(struct spi_device *spi) { struct wm8741_priv *wm8741; - int ret; + int ret, i; wm8741 = devm_kzalloc(&spi->dev, sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + wm8741->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8741); -- 1.7.10.4

1 1

[alsa-devel] ASoC: Multiplexing CODECs together
by Mike Looijmans 10 Sep '12

10 Sep '12

I'm using multiple TI codecs and an OMAP-L138. So far, I've been able to use a single codec. I've changed the davinci-mcasp.c code to respond to "set_tdm_slot", which I now use to tell it which serializers must be activated for capture and playback. This works fine now, any codec on any serializer port can be used independently now. I want to take the next step and have two codecs "mixed" together, so that two codecs form a 4-channel ADC device. They are all running on the same clocks. My idea so far is to create a "multiplex" codec driver, which exports mixer controls to tell it which real codecs to activate. No problem so far. Now when an application starts capture from the multiplex, it should call the "real" codecs (e.g. set_hw_params). This poses several problems, because the structs passed to the calls contain information that is irrelevant for the codec(s) that must actually do the work. Creating a multiplex codec, as opposed to say a dai, made sense to me because an application can "select" it. So my question is, Am I on the right track here? Or should I forget about the codec multiplex driver, because I have to write something completely? Or is this not the correct list to ask this kind of questions? In which case I'd be happy to learn where I might get some tips? Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) - (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. Ieder gebruik van de inhoud van het e-mail bericht, waaronder de daarbij behorende bijlagen, door een ander dan de geadresseerde is onrechtmatig jegens ons dan wel de eventueel in het e-mail bericht of de bijlagen voorkomende andere personen. TOPIC Embedded Systems is niet aansprakelijk voor enigerlei schade voortvloeiend uit het gebruik en/of acceptatie van dit e-mail bericht of de daarbij behorende bijlagen. The contents of this message, as well as any enclosures, are addressed personally to, and thus solely intended for the addressee. They may contain information regarding a third party. A recipient who is neither the addressee, nor empowered to receive this message on behalf of the addressee, is kindly requested to immediately inform the sender of receipt, and to destroy the message and the enclosures. Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

1 0

[alsa-devel] ASoC: fsi: tidyup patches
by Kuninori Morimoto 10 Sep '12

10 Sep '12

Hi Mark These are tidyup patches for fsi.c Kuninori Morimoto (3): ASoC: fsi: fixup pm_runtime_disable() timing on fsi_probe() ASoC: fsi: tidyup: remove un-necessary operation from fsi_probe() ASoC: fsi: convert to devm_xxx() sound/soc/sh/fsi.c | 28 +++++++++------------------- 1 file changed, 9 insertions(+), 19 deletions(-) Best regards --- Kuninori Morimoto

2 4

[alsa-devel] [PATCH 1/2] ASoC: sta32x: Move regulator acquisition to I2C probe
by Mark Brown 10 Sep '12

10 Sep '12

This is better style as it ensures we don't try to do the ASoC probe without required resources. Also convert to devm_ while we're at it, saving a bit of code, and fix a leak of enable on error. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sta32x.c | 32 +++++++++++++++----------------- 1 file changed, 15 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 51b7313..039597e 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -825,22 +825,11 @@ static int sta32x_probe(struct snd_soc_codec *codec) sta32x->codec = codec; sta32x->pdata = dev_get_platdata(codec->dev); - /* regulators */ - for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) - sta32x->supplies[i].supply = sta32x_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), - sta32x->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; - } - ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; + return ret; } /* Tell ASoC what kind of I/O to use to read the registers. ASoC will @@ -849,7 +838,7 @@ static int sta32x_probe(struct snd_soc_codec *codec) ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - return ret; + goto err; } /* Chip documentation explicitly requires that the reset values @@ -915,9 +904,8 @@ static int sta32x_probe(struct snd_soc_codec *codec) return 0; -err_get: - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); err: + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return ret; } @@ -928,7 +916,6 @@ static int sta32x_remove(struct snd_soc_codec *codec) sta32x_watchdog_stop(sta32x); sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; } @@ -966,13 +953,24 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct sta32x_priv *sta32x; - int ret; + int ret, i; sta32x = devm_kzalloc(&i2c->dev, sizeof(struct sta32x_priv), GFP_KERNEL); if (!sta32x) return -ENOMEM; + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, sta32x); ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); -- 1.7.10.4

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[alsa-devel] [PATCH 1/4] ASoC: wm8523: Convert to devm_kzalloc()
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8523.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 1c3ffb2..af2289e 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -516,7 +516,8 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, struct wm8523_priv *wm8523; int ret; - wm8523 = kzalloc(sizeof(struct wm8523_priv), GFP_KERNEL); + wm8523 = devm_kzalloc(&i2c->dev, sizeof(struct wm8523_priv), + GFP_KERNEL); if (wm8523 == NULL) return -ENOMEM; @@ -525,8 +526,7 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8523, &wm8523_dai, 1); - if (ret < 0) - kfree(wm8523); + return ret; } @@ -534,7 +534,6 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, static __devexit int wm8523_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.10.4

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Re: [alsa-devel] 3.6-rc Cirrus HDA reconfig oops...
by Takashi Iwai 10 Sep '12

10 Sep '12

At Sun, 9 Sep 2012 13:02:16 +0800, Daniel J Blueman wrote: > > With a non-specialised Cirrus HDA codec, it's readily possible to > cause a kernel oops as root with 3.6-rc kernels [1]. > > patch_cirrus.c: > > static int cs_init(struct hda_codec *codec) > { > struct cs_spec *spec = codec->spec; > > /* init_verb sequence for C0/C1/C2 errata*/ > snd_hda_sequence_write(codec, cs_errata_init_verbs); > > snd_hda_sequence_write(codec, cs_coef_init_verbs); > > if (spec->gpio_mask) { > snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, > spec->gpio_mask); > > Here, spec is NULL, so loading spec->gpio_mask causes a fatal > pagefault at address 0x180 (ie the offset of gpio_mask in the struct). > I was going to prepare a patch to guard for this, but since spec is > expected in so many places, there is a likely a behavioural issue > preventing a spec struct being generated. > > Let me know for further testing and debug. It's because of calling a leftover callback. The patch below should fix the issue. I'll queue it up to for-linus branch. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix Oops at codec reset/reconfig snd_hda_codec_reset() calls restore_pincfgs() where the codec is powered up again, which eventually tries to resume and initialize via the callbacks of the codec. However, it's the place just after codec free callback, thus no codec callbacks should be called after that. On a codec like CS4206, it results in Oops due to the access in init callback. This patch fixes the issue by clearing the codec callbacks properly after freeing codec. Reported-by: Daniel J Blueman <daniel(a)quora.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f25c24c..aa4cbca 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2353,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) } if (codec->patch_ops.free) codec->patch_ops.free(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; codec->spec = NULL; -- 1.7.11.5

1 1

[alsa-devel] [RFC update 0/2] dmaengine/ASoC: omap: Enable element mode in cyclic DMA
by Peter Ujfalusi 10 Sep '12

10 Sep '12

Hi Russell, Enable the element mode (thus allowing mono playback and probably unblocking OMAP1, OMAP2420) in OMAP dmaengine and omap-pcm. Janusz: would it be possible for you to test Russell's series plus this on OMAP1 to make sure that we do not broke it? Russell: we should wait for Janusz to test this on OMAP1. Based on the feedback we can plan on how to proceed with the dmaengine for OMAP audio. Regards, Peter --- Peter Ujfalusi (2): dmaengine: omap: Support for element mode in cyclic DMA ASoC: omap-pcm: Do not check DMA sync_mode drivers/dma/omap-dma.c | 5 ++++- sound/soc/omap/omap-pcm.c | 10 ---------- 2 files changed, 4 insertions(+), 11 deletions(-) -- 1.7.12

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[alsa-devel] [PATCH 1/2] ASoC: wm8983: Convert to devm_kzalloc()
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8983.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 367388f..57c33fc 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1085,7 +1085,7 @@ static int __devinit wm8983_spi_probe(struct spi_device *spi) struct wm8983_priv *wm8983; int ret; - wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL); + wm8983 = devm_kzalloc(&spi->dev, sizeof *wm8983, GFP_KERNEL); if (!wm8983) return -ENOMEM; @@ -1094,15 +1094,12 @@ static int __devinit wm8983_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8983, &wm8983_dai, 1); - if (ret < 0) - kfree(wm8983); return ret; } static int __devexit wm8983_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1123,7 +1120,7 @@ static __devinit int wm8983_i2c_probe(struct i2c_client *i2c, struct wm8983_priv *wm8983; int ret; - wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL); + wm8983 = devm_kzalloc(&i2c->dev, sizeof *wm8983, GFP_KERNEL); if (!wm8983) return -ENOMEM; @@ -1132,15 +1129,13 @@ static __devinit int wm8983_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8983, &wm8983_dai, 1); - if (ret < 0) - kfree(wm8983); + return ret; } static __devexit int wm8983_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.10.4

1 1

[alsa-devel] [PATCH] HDA-Intel: Fix wrong CTLINDEX entries for HBR Passthrough
by Wang Xingchao 10 Sep '12

10 Sep '12

>From Anssi Hannula: "The CTLINDEX=x values in /usr/share/alsa/cards/HDA-Intel.conf are off when you have S/PDIF output in the same card as HDMI, which is often the case on Intel. The workaround is to bump all the CTLINDEX=x values one step up in the HDA-Intel.pcm.hdmi.foo entries, i.e. CTLINDEX=0 => CTLINDEX=1 CTLINDEX=1 => CTLINDEX=2 etc, in the /usr/share/alsa/cards/HDA-Intel.conf." Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- src/conf/cards/HDA-Intel.conf | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/conf/cards/HDA-Intel.conf b/src/conf/cards/HDA-Intel.conf index d4f2667..7d6de8f 100644 --- a/src/conf/cards/HDA-Intel.conf +++ b/src/conf/cards/HDA-Intel.conf @@ -211,7 +211,7 @@ HDA-Intel.pcm.hdmi.0 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=3," - "CTLINDEX=0," + "CTLINDEX=1," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," @@ -234,7 +234,7 @@ HDA-Intel.pcm.hdmi.1 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=7," - "CTLINDEX=1," + "CTLINDEX=2," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," @@ -257,7 +257,7 @@ HDA-Intel.pcm.hdmi.2 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=8," - "CTLINDEX=2," + "CTLINDEX=3," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," @@ -280,7 +280,7 @@ HDA-Intel.pcm.hdmi.3 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=9," - "CTLINDEX=3," + "CTLINDEX=4," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," -- 1.7.9.5

3 2

[alsa-devel] ALSA release cycle
by David Henningsson 08 Sep '12

08 Sep '12

Hi, At Plumber's we discussed the ALSA release cycle. Our releases recently have been irregular, and the reasoning behind why a release was done at that time, has not been very obvious. IIRC, we kind of leaned towards releasing every six months. I don't remember if there was any consensus about whether to try to align this cycle to something else (e g Gnome, KDE, Fedora, Ubuntu, etc), or not. We also said that we should discuss this on the mailinglist as the ALSA release manager (Jaroslav Kysela) was not present during Plumber's. So this effectively is a mail to kick off that discussion. Any opinions? Also, as a side note (or perhaps proof of the problem!), it seems ALSA 1.0.26 was just released without even a notification on this mailinglist...? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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Re: [alsa-devel] How to define card specific pcm devices?
by Liam Girdwood 08 Sep '12

08 Sep '12

On Wed, 2012-09-05 at 12:24 +0100, Girdwood, Liam wrote: > On 5 September 2012 06:41, Tanu Kaskinen <tanuk(a)iki.fi> wrote: > On Wed, 2012-08-29 at 09:44 +0800, Raymond Yau wrote: > > strictly speaking your mic is not a true stereo mic and the > channel > > map is [left, null] instead of the normal map [left,right] > of line-in > > or stereo mic > > > I think specifying the channel map like you suggest would be a > good > solution. Supporting it would require some extra work in > PulseAudio, but > I think PulseAudio will anyway have to be extended to query > the channel > map information from UCM. > > Now the problem is that UCM doesn't provide the channel map > information > at all. Should the channel map be provided in the device > "Value" > section? Like this: > > SectionDevice."my-mono-mic" { > > ... > > Value { > # Does this value become redundant if > # the "CaptureChannelMap" value is added? > CaptureChannels "2" > I would expect to see a "1" here unless we cannot open the pcm in mono mode. > # Channel names match the constants in > # Takashi's proposed channel map API > CaptureChannelMap "FL, NA" > } > > } This seems fine and should be a simple patch for UCM too. Pulsaudio could use the map information/preference from UCM (like above) to configure the driver using Takashi's new channel map API. > > Opinions? (Liam added to CC - am I right in that you should be > CC'd in > all UCM related discussions?) Yes, please and Mark too (now on CC). Thanks Liam >

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[alsa-devel] sound/soc/soc-dapm.c:1023:4: error: implicit declaration of function 'regulator_allow_bypass'
by Fengguang Wu 08 Sep '12

08 Sep '12

[add the missing subject line..] On Sat, Sep 08, 2012 at 09:44:50AM +0800, Fengguang Wu wrote: > Hi Mark, > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > head: e8bb382aa74747f23e2903c15e48e9f06de5c60a > commit: 9ad4ba77112a3700129cdb142796a54680baa266 [100/103] ASoC: dapm: Allow regulators to bypass as well as disable when idle > config: x86_64-allmodconfig (attached as .config) > > All related error/warning messages: > > sound/soc/soc-dapm.c: In function 'dapm_regulator_event': > sound/soc/soc-dapm.c:1023:4: error: implicit declaration of function 'regulator_allow_bypass' [-Werror=implicit-function-declaration] > cc1: some warnings being treated as errors > > vim +1023 sound/soc/soc-dapm.c > 1018 { > 1019 int ret; > 1020 > 1021 if (SND_SOC_DAPM_EVENT_ON(event)) { > 1022 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { > > 1023 ret = regulator_allow_bypass(w->regulator, true); > 1024 if (ret != 0) > 1025 dev_warn(w->dapm->dev, > 1026 "Failed to bypass %s: %d\n", > 1027 w->name, ret); > 1028 } > > --- > 0-DAY kernel build testing backend Open Source Technology Centre > Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

1 0

Re: [alsa-devel] UAC2 clock selector for AudioBox 1818VSL
by louis＠museresearch.com 07 Sep '12

07 Sep '12

Hi Daniel, Thanks for the quick reply. I am able to set the clock on my Mac, though I do need to do further tests and make sure that the sync is actually reliable there. The setting is available from the built-in, standard OS X audio setup application "Audio MIDI Setup". There should be a S/PDIF clock signal going to the digital inputs on the device. Some of the other controls such as per-channel mute also appear to work. I've also attached the lsusb output (attachment stripped for the mailing list). There are some other details that may be relevant, but I didn't want to make my initial post too long/confusing because these extra details may end up just being a red herring: Each mixer element name (as seen in amixer) is prefixed with "AudioBox 1818 VSL Clock Selector" even if it is not a clock selector. I am confident though that the control I've been playing with is the clock selector because it lists the different clock types as options. I also see this when I plug the device in or load the ALSA driver: ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 (id 10) ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 (id 10) ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xb00, type = 4 ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 (id 11) ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xb00, type = 4 ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 (id 11) Amixer reports the control at numid=2 to be a boolean, but because of the naming bug I can't see what the control really is. I'm assuming this isn't relevant, but I'm including it here anyway. Louis > On 06.09.2012 22:20, louis(a)museresearch.com wrote: >> Hi! >> I'm attempting to get the "Clock Selector" mixer control to work with >> the PreSonus AudioBox 1818VSL (a UAC2 device). >> >> The control shows up in amixer as an enumerated value, but I don't seem to >> be able to set it (I've tried amixer, alsactl, and setting it >> programmatically). The possible values are correct (listing Internal Clock, S/PDIF Clock, and ADAT Clock as possible options); It just doesn't >> appear to take my setting, by which I mean amixer doesn't update the settings, I still have the occasional click when the sampling rates match >> over S/PDIF and I attempt to use external sync. No error is returned. >> >> After digging around USB specs and ALSA's usb/mixer.c and being unfamiliar >> with both the USB audio spec and the inner workings of ALSA, I thought it'd be best to ask the list about this: How would I check to see if ALSA >> is asking the device to change clock source at all? > > Hmm. A clock selector is handled like any other control entity on the hardware. Some thoughts: > > - Are you able to control other settings like the volume? > > - It's fairly common for devices to not accept a new clock source setting if they can't lock to the external source within some time. Did you connect anything to the digital inputs? > > - Are you able to control that setting with the native Mac OS X driver? (IOW, without installing a 3rd-party driver) > > Also, please send the output of "lsusb -v" for that device. > > > Daniel >

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[alsa-devel] [PATCH 1/4] ASoC: dapm: Add flags to regulator supplies
by Mark Brown 07 Sep '12

07 Sep '12

This will be used to enable additional control of the regulators. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++-- sound/soc/codecs/ab8500-codec.c | 8 ++++---- sound/soc/codecs/wm2200.c | 4 ++-- sound/soc/codecs/wm5100.c | 6 +++--- sound/soc/codecs/wm5102.c | 12 ++++++------ sound/soc/codecs/wm5110.c | 12 ++++++------ sound/soc/codecs/wm8996.c | 2 +- 7 files changed, 25 insertions(+), 24 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 07e2510..c96bf5a 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -244,10 +244,11 @@ struct device; { .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ .shift = wshift, .invert = winvert, .event = wevent, \ .event_flags = wflags} -#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \ +#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags) \ { .id = snd_soc_dapm_regulator_supply, .name = wname, \ .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ - .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .invert = wflags} /* dapm kcontrol types */ diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b783650..3f46bff 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -390,10 +390,10 @@ static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), /* Regulators */ - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0, 0), /* Power */ SND_SOC_DAPM_SUPPLY("Audio Power", diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 71debd0..efa93dbb 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1117,8 +1117,8 @@ SND_SOC_DAPM_SUPPLY("MICBIAS1", WM2200_MIC_BIAS_CTRL_1, WM2200_MICB1_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM2200_MIC_BIAS_CTRL_2, WM2200_MICB2_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20, 0), SND_SOC_DAPM_INPUT("IN1L"), SND_SOC_DAPM_INPUT("IN1R"), diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 33aa363..f4b327e 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -893,9 +893,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM5100_CLOCKING_3, WM5100_SYSCLK_ENA_SHIFT, 0, SND_SOC_DAPM_SUPPLY("ASYNCCLK", WM5100_CLOCKING_6, WM5100_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), SND_SOC_DAPM_SUPPLY("CP1", WM5100_HP_CHARGE_PUMP_1, WM5100_CP1_ENA_SHIFT, 0, NULL, 0), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 97223e4..3f5da8c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -312,12 +312,12 @@ SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0), SND_SOC_DAPM_SIGGEN("TONE"), SND_SOC_DAPM_SIGGEN("NOISE"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 94cd4f1..e7f8d17 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -359,12 +359,12 @@ SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0), SND_SOC_DAPM_SIGGEN("TONE"), SND_SOC_DAPM_SIGGEN("NOISE"), diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 00f183d..6dcb02c 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -931,7 +931,7 @@ SND_SOC_DAPM_INPUT("IN2RP"), SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), -- 1.7.10.4

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[alsa-devel] [PATCH v3 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 07 Sep '12

07 Sep '12

Hello, Generated on top of: git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap Changed since v2: - Added commit message to patch 2 (as Tero requested it) - Fixed patch 6 (dt: Add empty...) since the previous patch caused x86_64 build to fail due to missing struct before device_node. Changes since v1: - Get the MCLK frequencey from twl-core driver (via new API) - hs_extmute_disable_level parameter has been removed - empty of_find_node_by_name() in of.h for !CONFIG_OF builds Mark: the extmute GPIO handling (when it is used) remained in the codec driver for now. I can think more on how to add support for this type of mute control. If you can point me to other codec needing this it would help on designing something which is common enough for other users. As for now I have a separate series to add GPIO controlled output amps (hp/spk). I will send that to review soon. Currently this piece of code does not work when booted with device tree since I have done it on n900 which has the GPIO numbers as defines. I need to revisit the aproach on how to do it in a dynamic way. Introl mail from v1: The following series adds DT support for the twl4030 audio submodule which provides audio codec and vibra functionality. The MFD core driver is probed via DT, it will create the needed child devices based on the provided information in the DT blob. Child drivers (vibra, ASoC codec) will parse the core's node if needed to get the needed parameters for their configuration. In the ASoC codec driver the hs_extmute callback (which was used to toggle a GPIO line) has been removed. The codec driver will receive the GPIO number (if it is needed on the platform). If the series is OK (and no objections from the maintainers), it would be good if this can go via audio. Changed files are well contained within the twl4030-audio stack so I do not expect merge issues later. The series has been tested on BeagleBoard (with the McBSP DT series, and with the upcoming DT audio support for BeagleBoard). Regards, Peter --- Peter Ujfalusi (14): MFD: twl4030-audio: Clean up MODULE_* and platform_driver part MFD: twl4030-audio: Convert to use devm_kzalloc MFD: twl4030-audio: Rearange and clean-up the probe function MFD: twl-core: Add API to query the HFCLK rate MFD: twl4030-audio: Get audio MCLK via twl-core API instead of pdata dt: Add empty of_find_node_by_name() function MFD: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/MFD: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel .../devicetree/bindings/mfd/twl4030-audio.txt | 46 +++++++++ arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - drivers/input/misc/twl4030-vibra.c | 20 +++- drivers/mfd/twl-core.c | 32 ++++++ drivers/mfd/twl4030-audio.c | 107 ++++++++++++++------- include/linux/i2c/twl.h | 3 +- include/linux/of.h | 6 ++ sound/soc/codecs/twl4030.c | 103 ++++++++++++++++---- sound/soc/omap/zoom2.c | 4 - 10 files changed, 261 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt -- 1.7.12

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[alsa-devel] ALSA package release v1.0.26 (user space)
by Jaroslav Kysela 07 Sep '12

07 Sep '12

Hello all, I would like to notify you, that ALSA packages version 1.0.26 are available for download: ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/utils/alsa-utils-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.26.1.tar.bz2 ftp://ftp.alsa-project.org/pub/plugins/alsa-plugins-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/pyalsa/pyalsa-1.0.26.tar.bz2 The alsa-firmware and alsa-oss packages have no changes and they were not released this time. The changelog is available at this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.25_v1.0.26 The alsa-driver package will be released in next weeks once the driver repository with the build code and the mirrored kernel code will be reorganized to be more "GIT pull" friendly for those using the whole kernel tree as the development platform. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [alsa-jack-plugin] Feedback to modification proposal
by Merx, Thomas 07 Sep '12

07 Sep '12

Hello all, I would like to get some feedback here from you experts on the alsa-jack-plugin. In my project there is a jack-plugin that provides several jack input ports were different streams from different source applications shall be connectable. The source applications would provide the audio samples to jack via the alsa-jack-plugin. On top of that there is a control layer that has to connect the jack output ports of the alsa-jack-plugin with the input ports of my jack-plugin. The API of the control layer simply uses an ID for the source and an ID for the sink to know which jack output ports have to be connected to which input ports. The challenge I'm now facing is that the full name of the jack ports registered by the alsa-jack-plugin are changing from startup to startup because they include the process id of the playback application. Considering the case that there could be two instances of the same application and each of them providing a playback stream via the alsa-jack-plugin, it is challenging for the control layer to choose the correct stream for connecting it to my jack-plugin. So my main problem is that I cannot associate the IDs used for the control layer with the jack port names automatically created by the alsa-jack-plugin. A possible solution to my problem could be to add an additional entry in the asound.conf section where the jack plugin is defined to provide a name which shall be chosen as a prefix for the full jack port name, instead of the default naming scheme currently chosen. To have it backward compatible it could be implemented in a way that if the entry in the asound.conf is missing the current naming scheme is chosen and only if the new entry is set it will be used for the name to be registered at the jack server. What do you think about that? Does it make sense to have that functionality? Best regards Thomas Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Peter Gleissner, Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

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[alsa-devel] [RFC PATCH 1/11] expose sound device topology information
by Clemens Ladisch 07 Sep '12

07 Sep '12

Hi, these are some tentative patches to allow sound drivers to provide routing information to userspace. (So far, snd-usb-audio has a partial implementation; I haven't yet had time for snd-hda-intel. I haven't even looked at ASoC.) The media controller API <linux/media.h> allows prividing routing information, but its implementation does not quite fit the sound drivers: * it allows reconfiguration of links, but not in a way that would be useful for ALSA selector controls; * it allows drivers to access the entity graph and pipelines, which is not needed; * it is rather heavyweight. Therefore, these patches create their own implementation of the media controller API. In this implementation, all entity and link information is completely static, and more space-efficient. For simplicity, the ioctls hook into the ALSA control device instead of creating a new /dev/media* device. The separate implementation does not allow sharing of one media controller device in the case of combined audio/video devices. However, separate drivers would already be a problem for HDMI outputs (GPU and sound); it appears we might need some mechanism to connect the topologies of multiple media devices. TLVs (for jack entities, and for associating controls with entities) are not yet defined. Example output for my simple UA-1A device (not all information is decoded correctly): $ media-ctl -p -d /dev/snd/controlC5 Opening media device /dev/snd/controlC5 Enumerating entities Found 4 entities Enumerating pads and links Media controller API version 0.0.0 Media device information ------------------------ driver snd-usb-audio model EDIROL UA-1A serial bus info usb-0000:00:12.2-4.3 hw revision 0x101 driver version 0.0.0 Device topology - entity 1: USB Audio (1 pad, 1 link) type Node subtype ALSA device node name /dev/tty (should be hw:5,0,0) pad0: Source -> "OT3":0 [ENABLED,IMMUTABLE] - entity 3: OT3 (1 pad, 1 link) type Unknown subtype Unknown (should be Jack) pad0: Sink <- "USB Audio":0 [ENABLED,IMMUTABLE] - entity 4: IT4 (1 pad, 1 link) type Unknown subtype Unknown pad0: Source -> "USB Audio":0 [ENABLED,IMMUTABLE] - entity 7: USB Audio (1 pad, 1 link) type Node subtype ALSA device node name /dev/tty pad0: Sink <- "IT4":0 [ENABLED,IMMUTABLE] include/linux/media.h | 9 include/sound/core.h | 13 + include/sound/media.h | 116 ++++++++++ include/sound/pcm.h | 6 sound/core/Kconfig | 7 sound/core/Makefile | 1 sound/core/init.c | 4 sound/core/media.c | 425 ++++++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 46 ++++ sound/core/sound.c | 3 sound/pci/hda/hda_intel.c | 2 sound/usb/card.c | 2 sound/usb/mixer.c | 278 ++++++++++++++++++++++++ sound/usb/quirks.c | 4 sound/usb/stream.c | 16 - sound/usb/stream.h | 1 16 files changed, 925 insertions(+), 8 deletions(-) Regards, Clemens

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[alsa-devel] [RFC] Channel mapping API (take 2)
by Takashi Iwai 07 Sep '12

07 Sep '12

Hi, my proposal for channel mapping API seems accepted fairly well through discussions at Plumbers, so I continued to work on it and uploaded the updated version. The kernel part is found in sound-unstable tree topic/tlv-chmap branch git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git and the alsa-lib part is found in github tree topic/chmap branch git://git.github.com/tiwai/alsa-lib.git The updated parts are: - SNDRV_CHMAP_NA was newly added, indicating the channel is not available or silent. - The channel position value is masked in lower 16bit. The upper bits are used for more channel attributes as bit flags. #define SNDRV_CHMAP_POSITION_MASK 0xffff The phase inverted channel has this bit: #define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) And the driver-specific non-standard channel position has this: #define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) - The alsa-lib API functions use snd_pcm_chmap_query_t and snd_pcm_chmap_t instead of ambiguous integer arrays. ================================================================ /** the channel map header */ typedef struct snd_pcm_chmap { unsigned int channels; unsigned int pos[0]; } snd_pcm_chmap_t; /** the header of array items returned from snd_pcm_query_chmaps() */ typedef struct snd_pcm_chmap_query { enum snd_pcm_chmap_type type; snd_pcm_chmap_t map; } snd_pcm_chmap_query_t; snd_pcm_chmap_query_t **snd_pcm_query_chmaps(snd_pcm_t *pcm); void snd_pcm_free_chmaps(snd_pcm_chmap_query_t **maps); snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm); int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); ================================================================ The rest aren't so much changed. Only slight bug fixes. A big remaining question is whether the current kernel-side implementation is OK. We may add some PCM ops pointer instead of the current style using the direct control tlv/read/write override. But the kernel-side API can be changed even later, so I don't worry about it so much for now, as long as we can keep the ABI definition. thanks, Takashi

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[alsa-devel] [PATCH 0/3] HDMI HBR(High Bit Rate) Feature bug fix for Intel's Chips
by Wang Xingchao 07 Sep '12

07 Sep '12

As i have no A/V receiver handy and could not test HBR playback directly, i removed the "RFC" type until the patchset was tested and proved working well. The patchset was tested on Intel chips and get positive feedback. Quote from "Øyvind Kvålsvoll <oyvind(a)kvalsvoll.com>": "TrueHD and DTS-MA work fine from XBMC, 2-ch playback also works fine from XBMC. Channels are all mapped to the right speaker, all 7.1." Quote from alanwww1: "I tested the patches. They work perfectly. I tested it with both DTS Master Audio and Dolby Tru HD streams. Also tested with speaker-test. Channel mapping was right." From: http://forum.xbmc.org/showthread.php?tid=128298&pid=1184574#pid1184574 Maybe there's still potential bug and i will keep on track that. The idea of this patch comes from Anssi, Big credit to him at first! Thanks the guys from XBMC forum which help test the patches. For people interested to test the patch, please remember to apply another change in alsa-lib side, you can refer to the patch detail from XBMC forum: http://forum.xbmc.org/showthread.php?tid=128298&pid=1178776#pid1178776 I will send that patch to alsa-lib mail-list later. Changelog: * do not touch channel_allocation priority order, use new api to get correct channels number * use spdif->status to check audio type, pcm or non-pcm * create new non_pcm_mapping[] for non_pcm audio in seperate block * add non_pcm hdmi_spec_per_cvt to remember previous audio type, used to determine whether need a channel mapping update during the pcm/non-pcm stream switch with same channel number. Wang Xingchao (3): ALSA: HDMI - Fix channel_allocation array wrong order ALSA: HDMI - Enable HBR feature on Intel chips ALSA: HDMI - Setup channel mapping for non_pcm audio sound/pci/hda/patch_hdmi.c | 56 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 51 insertions(+), 5 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] Should codec suspend when no paths are marked to be active across a suspend action?
by Mark Brown 07 Sep '12

07 Sep '12

On Thu, Sep 06, 2012 at 12:55:00PM -0700, Taylor Hutt wrote: > Shouldn't the codec be suspended if there are no paths set to remain > active across the suspend? Yes. If you're using an analogue bypass path there's a very recent commit in -next from me which should help, previously we weren't ever really suspending bypass paths at all (even before the changes to keep some paths active) though we might end up doing so by accident. Nobody noticed since essentially nobody uses analogue bypass any more.

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[alsa-devel] UAC2 clock selector for AudioBox 1818VSL
by louis＠museresearch.com 06 Sep '12

06 Sep '12

Hi! I'm attempting to get the "Clock Selector" mixer control to work with the PreSonus AudioBox 1818VSL (a UAC2 device). The control shows up in amixer as an enumerated value, but I don't seem to be able to set it (I've tried amixer, alsactl, and setting it programmatically). The possible values are correct (listing Internal Clock, S/PDIF Clock, and ADAT Clock as possible options); It just doesn't appear to take my setting, by which I mean amixer doesn't update the settings, I still have the occasional click when the sampling rates match over S/PDIF and I attempt to use external sync. No error is returned. After digging around USB specs and ALSA's usb/mixer.c and being unfamiliar with both the USB audio spec and the inner workings of ALSA, I thought it'd be best to ask the list about this: How would I check to see if ALSA is asking the device to change clock source at all? -Louis

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Re: [alsa-devel] [PATCH 6/9] ARM: OMAP: Move gpio.h to include/linux/platform_data
by Tony Lindgren 06 Sep '12

06 Sep '12

* Mohammed, Afzal <afzal(a)ti.com> [120903 03:15]: > Hi, > > On Fri, Aug 31, 2012 at 06:22:34, Tony Lindgren wrote: > > This way we can remove includes of plat/gpio.h which won't work > > with the single zImage support. > > > > Note that we also remove the cpu_class_is_omap2() check > > in gpio-omap.c as the drivers should not call it as we need to > > make it local to arch/arm/mach-omap2 for single zImage support. > > > > While at it, arrange the related includes in the standard way. > > > > Cc: Grant Likely <grant.likely(a)secretlab.ca> > > Cc: Linus Walleij <linus.walleij(a)linaro.org> > > Cc: linux-mtd(a)lists.infradead.org > > Cc: alsa-devel(a)alsa-project.org > > Acked-by: Tony Lindgren <tony(a)atomide.com> > > I think there is only one Tony ;) Heh thanks I'll tell myself to replace it with Signed-off-by :) Tony

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Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 06 Sep '12

06 Sep '12

At Thu, 6 Sep 2012 11:43:48 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 10:21 +0200, Takashi Iwai wrote: > > At Thu, 06 Sep 2012 09:35:26 +0200, > > Takashi Iwai wrote: > > > > In short, a patch like below may fix the issue (note: completely > > untested!) > > No it doesn't, unfortunately... OK, I start tracking down the problem a bit more deeply now. The issue happens when the first two URBs are passed to retire_playback_urb(). These are URBs filled before start_endpoints() are set, so they contain actually zero size. Even though these are a sort of dummy packets, the driver still tries to check with the queued delay account, and gives bogus errors. So, essentially the messages are harmless and nothing to worry too much, but surely it doesn't look sexy. The patch below should fix the problem. Please give it a try. thanks, Takashi === From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: usb-audio: Fix bogus error messages for delay accounting The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by: Markus Trippelsdorf <markus(a)trippelsdorf.de> Cc: <stable(a)vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index fd5e982..f782ce1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1140,6 +1140,12 @@ static void retire_playback_urb(struct snd_usb_substream *subs, int processed = urb->transfer_buffer_length / stride; int est_delay; + /* ignore the delay accounting when procssed=0 is given, i.e. + * silent payloads are procssed before handling the actual data + */ + if (!processed) + return; + spin_lock_irqsave(&subs->lock, flags); est_delay = snd_usb_pcm_delay(subs, runtime->rate); /* update delay with exact number of samples played */ -- 1.7.11.5

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[alsa-devel] [PATCH v2] ASoC: samsung dma - Don't indicate support for pause/resume.
by Dylan Reid 06 Sep '12

06 Sep '12

The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..b70964e 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | @@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: prtd->state &= ~ST_RUNNING; prtd->params->ops->stop(prtd->params->ch); break; -- 1.7.12.146.g16d26b1

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[alsa-devel] [PATCH Resend2] ASoC: imx-mc13783: use defines instead of numerical address of register
by Gaëtan Carlier 06 Sep '12

06 Sep '12

This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++-------------------- 1 files changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c0..bbfa553 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- 1.7.7.4

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Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 06 Sep '12

06 Sep '12

At Thu, 6 Sep 2012 09:17:57 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 09:08 +0200, Daniel Mack wrote: > > On 06.09.2012 08:53, Markus Trippelsdorf wrote: > > > On 2012.09.06 at 08:48 +0200, Takashi Iwai wrote: > > >> At Thu, 06 Sep 2012 08:33:30 +0200, > > >> Daniel Mack wrote: > > >>> > > >>> On 06.09.2012 08:02, Markus Trippelsdorf wrote: > > >>>> On 2012.09.04 at 16:40 +0200, Takashi Iwai wrote: > > >>>>> ---------------------------------------------------------------- > > >>>>> Sound fixes for 3.6-rc5 > > >>>>> > > >>>>> There are nothing scaring, contains only small fixes for HD-audio and > > >>>>> USB-audio: > > >>>>> - EPSS regression fix and GPIO fix for HD-audio IDT codecs > > >>>>> - A series of USB-audio regression fixes that are found since 3.5 kernel > > >>>>> > > >>>>> ---------------------------------------------------------------- > > >>>>> Daniel Mack (4): > > >>>>> ALSA: snd-usb: Fix URB cancellation at stream start > > >>>>> ALSA: snd-usb: restore delay information > > >>>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > >>>> The commit fbcfbf5f above causes the following lines to be printed > > >>>> whenever I start a new song: > > >>> > > >>> Copied Pierre-Louis Bossart - he wrote the code in 294c4fb8 which this > > >>> patch (fbcfbf5f) brings back now. > > >>> > > >>>> delay: estimated 0, actual 352 > > >>>> delay: estimated 353, actual 705 > > >>>> > > >>>> (44.1 * 8 = 352.8) > > >>>> > > >>>> This happens with an USB-DAC that identifies itself as "C-Media USB > > >>>> Headphone Set". > > >>> > > >>> And you didn't you see these lines with 3.4? > > >> > > >> Maybe the difference of start condition? > > >> > > >> Markus, does the patch below fix anything? > > > > > > Unfortunately no. > > > However reverting the following fixes the problem: > > > > > > commit 245baf983cc39524cce39c24d01b276e6e653c9e > > > Author: Daniel Mack <zonque(a)gmail.com> > > > Date: Thu Aug 30 18:52:30 2012 +0200 > > > > > > ALSA: snd-usb: fix calls to next_packet_size > > > > > > > No, this one certainly fixes a problem and does the right thing by > > restoring the original code. > > > > If you wouldn't state that you didn't see the same effect with 3.4(!), > > before the refactoring done in 3.5, I would believe the device is simply > > slightly off in its feedback rate and the tighter delay code complains > > about it while compensating, just as it did before. > > > > Are there any more than these two lines? And is audio working at all? Is > > it distorted in any way? > > There are only these two lines (printed whenever sound starts). Audio is > working just fine with no distortions. > > I did see similar lines before when the system load was very high > (happend during "make check" when building glibc). > > Here is what Pierre-Louis wrote in November 2011: > > Â»This was supposed to be an informational message, I thought it was only > enabled for debug. Regular users don't really need to know.Â« I guess the problem is that the new endpoint scheme doesn't count the last_delay update unless the stream is triggered. In the old code, retire_playback_urb is always called even before the trigger(START) is set. And, there retire_playback_urb() does nothing but updating the delay information. In the new code, retire_playback_urb is set only at snd_usb_substream_playback_trigger(). Thus at the very first shot, the delay account got confused. Takashi

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[alsa-devel] Multiple speaker jacks (on an IDT codec)
by David Henningsson 06 Sep '12

06 Sep '12

Hi, I'm having a problem with a machine where the parser creates indexed speaker controls. For reference, codecs/idt92hd206-ecs-gf7100pvt shows the same problem. I could use some input on how to fix it in the best way. My first thought was that "[Jack] Speaker" is stupid in itself, and could be fixed like this: diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4f7d2df..8e32ee5 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -171,6 +171,10 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, if (conn == AC_JACK_PORT_FIXED) dev = AC_JACK_SPEAKER; } + else if (dev == AC_JACK_SPEAKER) { + if (conn == AC_JACK_PORT_COMPLEX) + dev = AC_JACK_LINE_OUT; + } switch (dev) { case AC_JACK_LINE_OUT: ...but changing generic code is always a bit scary and regression prone (even though I could not think of a good counterexample where that would happen), so not sure if that's the best idea. Next attempt is to modify the control creating function for IDT: diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9db3056..c16d0f8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3226,9 +3226,12 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, idx = i; break; case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - idx = i; - break; + if (num_outs <= 1) { + name = "Speaker"; + idx = i; + break; + } + /* Fall through in case of multi speaker outs */ default: name = chname[i]; idx = 0; ...since the current behaviour is inconsistent anyway - you get indices for 1 and 3, but "Center"/"LFE" for what should otherwise have been index 2. Another way would to just write a pin fix to turn these speakers into line outs, which is what they really are anyway. What do you think? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Add SND_SOC_DAIFMT_CONT option for snd_soc_set_fmt()
by 박상수 06 Sep '12

06 Sep '12

SND_SOC_DAIFMT_CONT option is now needed because of below commit: ASoC: Allow DAI formats to be specified in the dai_link Otherwise, s3c_pcm_set_fmt() returns -EINVAL. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/smdk_wm8994pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 77ecba9..cb88cbc 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -77,6 +77,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, /* Set the cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_CONT | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) -- 1.7.1

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[alsa-devel] radeon multi-channel lpcm
by Pierre Ossman 06 Sep '12

06 Sep '12

Hi, I figured I'd have a go at finishing up support for multi-channel lcpm over HDMI on radeon cards. I read a post from the AMD guys stating that the GPU portions were already in place, and it was the audio pieces that were missing. Unfortunately I was unable to figure out exactly what it is that remains to be done. I poked around in patch_hdmi.c, and concluded that the generic code is what's being used for modern radeon cards. The card exposes two widgets (and output widget and a pin complex). But both widgets indicate through their capability registers that they only support stereo. I don't know enough about HDA to have any clue where to go from here. Should the capability field be ignored for HDMI? Or does it need some coaxing to allow more than stereo? The ELD stuff isn't properly updated from the radeon driver, but the code seems to suggest it is just for hints, and not critical. Rgds -- -- Pierre Ossman WARNING: This correspondence is being monitored by FRA, a Swedish intelligence agency. Make sure your server uses encryption for SMTP traffic and consider using PGP for end-to-end encryption.

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Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Daniel Mack 06 Sep '12

06 Sep '12

On 06.09.2012 08:53, Markus Trippelsdorf wrote: > On 2012.09.06 at 08:48 +0200, Takashi Iwai wrote: >> At Thu, 06 Sep 2012 08:33:30 +0200, >> Daniel Mack wrote: >>> >>> On 06.09.2012 08:02, Markus Trippelsdorf wrote: >>>> On 2012.09.04 at 16:40 +0200, Takashi Iwai wrote: >>>>> ---------------------------------------------------------------- >>>>> Sound fixes for 3.6-rc5 >>>>> >>>>> There are nothing scaring, contains only small fixes for HD-audio and >>>>> USB-audio: >>>>> - EPSS regression fix and GPIO fix for HD-audio IDT codecs >>>>> - A series of USB-audio regression fixes that are found since 3.5 kernel >>>>> >>>>> ---------------------------------------------------------------- >>>>> Daniel Mack (4): >>>>> ALSA: snd-usb: Fix URB cancellation at stream start >>>>> ALSA: snd-usb: restore delay information >>>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ >>>> The commit fbcfbf5f above causes the following lines to be printed >>>> whenever I start a new song: >>> >>> Copied Pierre-Louis Bossart - he wrote the code in 294c4fb8 which this >>> patch (fbcfbf5f) brings back now. >>> >>>> delay: estimated 0, actual 352 >>>> delay: estimated 353, actual 705 >>>> >>>> (44.1 * 8 = 352.8) >>>> >>>> This happens with an USB-DAC that identifies itself as "C-Media USB >>>> Headphone Set". >>> >>> And you didn't you see these lines with 3.4? >> >> Maybe the difference of start condition? >> >> Markus, does the patch below fix anything? > > Unfortunately no. > However reverting the following fixes the problem: > > commit 245baf983cc39524cce39c24d01b276e6e653c9e > Author: Daniel Mack <zonque(a)gmail.com> > Date: Thu Aug 30 18:52:30 2012 +0200 > > ALSA: snd-usb: fix calls to next_packet_size > No, this one certainly fixes a problem and does the right thing by restoring the original code. If you wouldn't state that you didn't see the same effect with 3.4(!), before the refactoring done in 3.5, I would believe the device is simply slightly off in its feedback rate and the tighter delay code complains about it while compensating, just as it did before. Are there any more than these two lines? And is audio working at all? Is it distorted in any way? Pierre-Louis, could you check whether I did the right thing when I ported over your delay bits to the new endpoint logic? Maybe I'm missing something here, but I currently don't see it. Thanks, Daniel

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Re: [alsa-devel] [PATCH 2/4] ARM: OMAP: Split plat/hardware.h, introduce local hardware.h and soc.h for omap2+
by Mark Brown 06 Sep '12

06 Sep '12

On Fri, Aug 31, 2012 at 05:26:22PM -0700, Tony Lindgren wrote: > As the plat and mach includes need to disappear for single zImage work, > we need to remove plat/hardware.h. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> There's a topic/omap branch in ASoC which you can pull in to help reduce conflicts, or I could merge this there if that's sensible.

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[alsa-devel] [PATCH Resend] ASoC: imx-mc13783: use defines instead of numerical address of register
by Gaëtan Carlier 06 Sep '12

06 Sep '12

Forgot to cc Mark Brown, sorry. This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++-------------------- 1 files changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c0..bbfa553 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- 1.7.7.4

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Re: [alsa-devel] (no subject)
by Daniel Mack 06 Sep '12

06 Sep '12

On 06.09.2012 08:02, Markus Trippelsdorf wrote: > On 2012.09.04 at 16:40 +0200, Takashi Iwai wrote: >> ---------------------------------------------------------------- >> Sound fixes for 3.6-rc5 >> >> There are nothing scaring, contains only small fixes for HD-audio and >> USB-audio: >> - EPSS regression fix and GPIO fix for HD-audio IDT codecs >> - A series of USB-audio regression fixes that are found since 3.5 kernel >> >> ---------------------------------------------------------------- >> Daniel Mack (4): >> ALSA: snd-usb: Fix URB cancellation at stream start >> ALSA: snd-usb: restore delay information > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > The commit fbcfbf5f above causes the following lines to be printed > whenever I start a new song: Copied Pierre-Louis Bossart - he wrote the code in 294c4fb8 which this patch (fbcfbf5f) brings back now. > delay: estimated 0, actual 352 > delay: estimated 353, actual 705 > > (44.1 * 8 = 352.8) > > This happens with an USB-DAC that identifies itself as "C-Media USB > Headphone Set". And you didn't you see these lines with 3.4? Daniel

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[alsa-devel] About joining codecs together
by Mike Looijmans 06 Sep '12

06 Sep '12

I have multiple codecs (TLV320AIC3256) connected to the McASP of an OMAP L138. I've adapted the drivers a bit so they work with my 2.6.37 kernel and the custom board (similar to a DA850). Currently, I can only use one codec at a time. The mcasp is set up in the board config, and there it determines which AXR lines are active. Each codec is connected to another line. They run on the same clock lines. Question: - How can I forward information from the codec to the mcasp driver, to inform which AXR channel to enable? There seems to be no information exchange between them. When codec 1 is running, the mcasp driver must record from AXR1, and when codec2 is running, AXR2 and so on. What I ultimately want to do is use two (or more) codecs for multichannel recording, so that I can use two codecs for a 4-channel recording. For the McASP side, this would simply mean enabling both AXR inputs (DAVINCI_MCASP_PDIR_REG) and then starting both codecs. Questions: - How does this work in user space? There are two codecs, which are to be configured independently, but there will be only one audio stream coming out. I was thinking to create a sort of "multiswitch", a codec that by itself doesn't really do anything, but presents itself as a single codec with lots of channels. The actual commands will be delivered to the appropriate "real" codecs, so that you can record from the "switch", which then in turn enables two codecs and sets up the DAI to record both streams. Would that be possible at all? What are the alternatives? Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) - (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. Ieder gebruik van de inhoud van het e-mail bericht, waaronder de daarbij behorende bijlagen, door een ander dan de geadresseerde is onrechtmatig jegens ons dan wel de eventueel in het e-mail bericht of de bijlagen voorkomende andere personen. TOPIC Embedded Systems is niet aansprakelijk voor enigerlei schade voortvloeiend uit het gebruik en/of acceptatie van dit e-mail bericht of de daarbij behorende bijlagen. The contents of this message, as well as any enclosures, are addressed personally to, and thus solely intended for the addressee. They may contain information regarding a third party. A recipient who is neither the addressee, nor empowered to receive this message on behalf of the addressee, is kindly requested to immediately inform the sender of receipt, and to destroy the message and the enclosures. Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

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[alsa-devel] [PATCH 1/1] [ASoC] Fix v2: Revert 'ASoC: imx-ssi: Remove mono support'.
by Gaëtan Carlier 06 Sep '12

06 Sep '12

When patch of Javier Martin is applied on linux-next-20120824 the patch failed because comment line is missing. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/fsl/imx-ssi.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 7074ae6..3c520c4 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,14 +380,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- 1.7.7.4

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[alsa-devel] [PATCH] ASoC: Revert 'ASoC: imx-ssi: Remove mono support'
by Fabio Estevam 06 Sep '12

06 Sep '12

From: Javier Martin <javier.martin(a)vista-silicon.com> Revert 0865a75(ASoC: imx-ssi: Remove mono support). The bug this patch is meant to solve doesn't occur in Visstrim_M10 boards. Furthermore, after applying this patch sound in Visstrim_M10 is played at slower rates. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/fsl/imx-ssi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 7074ae6..3c520c4 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,14 +380,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- 1.7.9.5

2 1

[alsa-devel] [PATCH -next] soc: fix wm0010.c printk format warning
by Randy Dunlap 06 Sep '12

06 Sep '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix printk format warning: sound/soc/codecs/wm0010.c:411:2: warning: format '%d' expects type 'int', but argument 4 has type 'size_t' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20120905.orig/sound/soc/codecs/wm0010.c +++ linux-next-20120905/sound/soc/codecs/wm0010.c @@ -408,7 +408,7 @@ static int wm0010_boot(struct snd_soc_co wm0010->state = WM0010_BOOTROM; spin_unlock_irqrestore(&wm0010->irq_lock, flags); - dev_dbg(codec->dev, "Downloading %d byte stage 2 loader\n", fw->size); + dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ img = kzalloc(fw->size, GFP_KERNEL);

2 3

[alsa-devel] [PATCH V2] ASoC: Davinci: McASP: add support new McASP IP Variant
by Hebbar, Gururaja 06 Sep '12

06 Sep '12

The OMAP2+ variant of McASP is different from Davinci variant w.r.to some register offset. Changes - Add new MCASP_VERSION_3 to identify new variant. New DT compatible "ti,omap2-mcasp-audio" to identify version 3 controller. - The register offsets are handled depending on the version. Note: DMA parameters (dma fifo offset) are not updated and will be done later. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- Changes in V2: Use "switch" instead of "if" statement to allow future revisions :100644 100644 e6148ec... 374e145... M Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt :100644 100644 79c26aa... d0c5825... M include/linux/platform_data/davinci_asp.h :100644 100644 c3eae1d... 714e51e... M sound/soc/davinci/davinci-mcasp.c .../bindings/sound/davinci-mcasp-audio.txt | 1 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 86 +++++++++++++++++--- 3 files changed, 75 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index e6148ec..374e145 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,6 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) - reg : Should contain McASP registers offset and length - interrupts : Interrupt number for McASP diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 79c26aa..d0c5825 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -87,6 +87,7 @@ struct snd_platform_data { enum { MCASP_VERSION_1 = 0, /* DM646x */ MCASP_VERSION_2, /* DA8xx/OMAPL1x */ + MCASP_VERSION_3, /* TI81xx/AM33xx */ }; enum mcbsp_clk_input_pin { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c3eae1d..714e51e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -111,6 +111,10 @@ #define DAVINCI_MCASP_WFIFOSTS (0x1014) #define DAVINCI_MCASP_RFIFOCTL (0x1018) #define DAVINCI_MCASP_RFIFOSTS (0x101C) +#define MCASP_VER3_WFIFOCTL (0x1000) +#define MCASP_VER3_WFIFOSTS (0x1004) +#define MCASP_VER3_RFIFOCTL (0x1008) +#define MCASP_VER3_RFIFOSTS (0x100C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -384,18 +388,36 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } } mcasp_start_tx(dev); } else { if (dev->rxnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } mcasp_start_rx(dev); } @@ -416,14 +438,31 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + if (dev->txnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_tx(dev); } else { - if (dev->rxnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + if (dev->rxnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_rx(dev); } } @@ -622,20 +661,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + tx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + } } if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { if (dev->rxnumevt * rx_ser > 64) dev->rxnumevt = 1; - - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + rx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + } } } @@ -874,6 +930,10 @@ static const struct of_device_id mcasp_dt_ids[] = { .compatible = "ti,da830-mcasp-audio", .data = (void *)MCASP_VERSION_2, }, + { + .compatible = "ti,omap2-mcasp-audio", + .data = (void *)MCASP_VERSION_3, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); -- 1.7.1

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Re: [alsa-devel] [ASoC] Fix: Revert 'ASoC: imx-ssi: Remove mono support'.
by Mark Brown 06 Sep '12

06 Sep '12

On Tue, Sep 04, 2012 at 11:09:20AM +0200, javier Martin wrote: > could you pick up the fixed patch sent by Gaëtan? It's mangled, can someone resend please (ideally with an appropriate subject line and so on).

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Re: [alsa-devel] [PATCH] ASoC: mc13783: Remove mono support
by Mark Brown 06 Sep '12

06 Sep '12

On Mon, Sep 03, 2012 at 01:04:13PM -0300, Fabio Estevam wrote: > Playing a mono track on a mc13783 codec results in incorrect playback rate. > > Remove mono support so that a mono track can be played correctly. Applied, thanks.

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[alsa-devel] [PATCH] ASoC: Remove unused 'saved_value' field from snd_soc_dapm_widget struct
by Lars-Peter Clausen 06 Sep '12

06 Sep '12

The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove current PGA control handling"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index abe373d..0a15537 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,7 +510,6 @@ struct snd_soc_dapm_widget { /* dapm control */ int reg; /* negative reg = no direct dapm */ unsigned char shift; /* bits to shift */ - unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: imx-mc13783: use defines instead of numerical address of register
by Gaëtan Carlier 06 Sep '12

06 Sep '12

This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++-------------------- 1 files changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c0..bbfa553 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- 1.7.7.4

3 2

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Fix compilation error due to leftover code
by Peter Ujfalusi 06 Sep '12

06 Sep '12

Part of commit (which patches sound/soc/omap/mcbsp.c file): 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code since the tree where it has been applied did not had the earlier patch: d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration which changed code around omap_mcbsp_6pin_src_mux(). Because of the missing part from 8fef626 the sound/soc/omap/mcbsp.c does not compile in linux-next. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, This patch has been generated against linux-next as of today. The reason for this is that that is the only place where we have the upstream patch (in 3.6-rc3): d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration and the 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code commit in the same branch. When you applied 8fef626 you mentioned that it did not applied cleanly. The reason was that the branch did not had commit d0db84e applied prior to the series: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-August/054668.html Now I can not find the d0db84e commit in the asoc tree (not in for-3.6, for-3.7 and not in for-next branch) anymore. I don't really know what has happened, but as of now this patch is needed to fix the OMAP audio in linux-next. I have also prepared a branch with the OMAP patches [1]: topic/omap rebased on top of today's mainline and fixed up the 8fef626 to it's original content. But I think it is already late to revert the OMAP McBSP patches from for-next and from for-3.7 branches... [1] git://gitorious.org/omap-audio/linux-audio.git peter/topic/for-mark/omap-3.7 CC-ing linux-next as well. Regards, Peter sound/soc/omap/mcbsp.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 935ccf6..bc06175 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -762,37 +762,6 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) } -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) -{ - const char *signal, *src; - - if (!mcbsp->pdata->mux_signal) - return -EINVAL; - - switch (mux) { - case CLKR_SRC_CLKR: - signal = "clkr"; - src = "clkr"; - break; - case CLKR_SRC_CLKX: - signal = "clkr"; - src = "clkx"; - break; - case FSR_SRC_FSR: - signal = "fsr"; - src = "fsr"; - break; - case FSR_SRC_FSX: - signal = "fsr"; - src = "fsx"; - break; - default: - return -EINVAL; - } - - return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); -} - #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) #define THRESHOLD_PROP_BUILDER(prop) \ -- 1.7.12

3 4

[alsa-devel] [PATCH] ASoC: arizona: Fix typo in 44.1kHz rates
by Mark Brown 06 Sep '12

06 Sep '12

From: Heather Lomond <hlomond(a)opensource.wolfsonmicro.com> Signed-off-by: Heather Lomond <hlomond(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9caca..1cf7a32 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = { 940800, 1411200, 1881600, - 2882400, + 2822400, 3763200, 5644800, 7526400, -- 1.7.10.4

1 0

[alsa-devel] [asoc:for-3.7 95/95] sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT
by Fengguang Wu 06 Sep '12

06 Sep '12

[add more CC on request of Dimitris] Hi Dimitris, Mark, FYI, there are new coccinelle warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 head: e3523e01869da20fdd12ffd19ae1df7bf492650e commit: e3523e01869da20fdd12ffd19ae1df7bf492650e [95/95] ASoC: wm0010: Add initial wm0010 DSP driver All coccinelle warnings: + sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT -- + sound/soc/codecs/wm0010.c:660:1-7: preceding lock on line 359 vim +850 sound/soc/codecs/wm0010.c 847 trigger = IRQF_TRIGGER_FALLING; 848 trigger |= IRQF_ONESHOT; 849 > 850 ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, 851 "wm0010", wm0010); 852 if (ret) 853 dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", Please consider folding the attached diff :-) --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

3 10

Re: [alsa-devel] [PATCH] ASoC: wm0010: Fix warning, use format %zu for type size_t
by Mark Brown 06 Sep '12

06 Sep '12

On Wed, Sep 05, 2012 at 10:22:24PM +0200, Emil Goode wrote: > Fix warning by using format specifier %zu for type size_t > > Sparse warning: > sound/soc/codecs/wm0010.c:411:2: warning: > format ‘%d’ expects argument of type ‘int’, > but argument 4 has type ‘size_t’ [-Wformat] Applied, thanks.

1 0

[alsa-devel] [PATCH V2] ASoC: tegra: move platform data header
by Stephen Warren 06 Sep '12

06 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: * Remove stale filename from comment. * Add include guards --- .../sound/tegra_wm8903.h | 7 +++++-- sound/soc/tegra/tegra_wm8903.c | 3 +-- 2 files changed, 6 insertions(+), 4 deletions(-) rename arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h => include/sound/tegra_wm8903.h (89%) diff --git a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h b/include/sound/tegra_wm8903.h similarity index 89% rename from arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h rename to include/sound/tegra_wm8903.h index 9d29334..57b202e 100644 --- a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h +++ b/include/sound/tegra_wm8903.h @@ -1,6 +1,4 @@ /* - * arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h - * * Copyright 2011 NVIDIA, Inc. * * This software is licensed under the terms of the GNU General Public @@ -14,6 +12,9 @@ * */ +#ifndef __SOUND_TEGRA_WM38903_H +#define __SOUND_TEGRA_WM38903_H + struct tegra_wm8903_platform_data { int gpio_spkr_en; int gpio_hp_det; @@ -21,3 +22,5 @@ struct tegra_wm8903_platform_data { int gpio_int_mic_en; int gpio_ext_mic_en; }; + +#endif diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index d4f14e4..cee13b7 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,13 +34,12 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <mach/tegra_wm8903_pdata.h> - #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tegra_wm8903.h> #include "../codecs/wm8903.h" -- 1.7.0.4

2 1

[alsa-devel] [PATCH] hda-emu: Add support for new pm_notify callback format
by David Henningsson 05 Sep '12

05 Sep '12

The format of the pm_notify callback has recently changed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 24 ++++++++++++++++++++++++ hda-emu.c | 12 ++++++++++++ 2 files changed, 36 insertions(+) diff --git a/configure.ac b/configure.ac index 18118f3..43db88f 100644 --- a/configure.ac +++ b/configure.ac @@ -222,6 +222,30 @@ if test "$OLD_HDA_CMD" = "1"; then AC_DEFINE(OLD_HDA_CMD) fi +AC_MSG_CHECKING(for new pm_notify callback) +ac_save_CFLAGS="$CFLAGS" +CFLAGS="$CFLAGS -Iinclude" +HAVE_NEW_PM_NOTIFY= +AC_TRY_COMPILE([ +#include "hda-types.h" +#include "hda-log.h" +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include "kernel/hda_codec.h" +],[ + extern struct hda_bus_ops *ops; + ops->pm_notify(NULL, 0); +], + [AC_MSG_RESULT(yes); HAVE_NEW_PM_NOTIFY="1"], + [AC_MSG_RESULT(no)], + [AC_MSG_RESULT(unknown)] +) +CFLAGS=$ac_save_CFLAGS +if test "$HAVE_NEW_PM_NOTIFY" = "1"; then + AC_DEFINE(HAVE_NEW_PM_NOTIFY) +fi + AC_MSG_CHECKING(for attach_pcm bus ops) ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS -Iinclude" diff --git a/hda-emu.c b/hda-emu.c index b4bd8a9..2a2b3d1 100644 --- a/hda-emu.c +++ b/hda-emu.c @@ -776,6 +776,13 @@ static int azx_pcm_create(struct hda_codec *codec) * power management */ +#ifdef HAVE_NEW_PM_NOTIFY +static void new_pm_notify(struct hda_bus *bus, bool power_up) +{ + hda_log(HDA_LOG_INFO, "PM-Notified, power_up = %d\n", power_up); +} +#else + static void pm_notify(struct hda_bus *bus) { hda_log(HDA_LOG_INFO, "PM-Notified\n"); @@ -787,6 +794,7 @@ static void old_pm_notify(struct hda_codec *codec) pm_notify(codec->bus); } #endif +#endif /* HAVE_NEW_PM_NOTIFY */ /* @@ -1112,7 +1120,11 @@ int main(int argc, char **argv) #ifdef HAVE_HDA_ATTACH_PCM temp.ops.attach_pcm = attach_pcm; #endif +#ifdef HAVE_NEW_PM_NOTIFY + temp.ops.pm_notify = new_pm_notify; +#else temp.ops.pm_notify = pm_notify; +#endif #endif /* OLD_HDA_CMD */ gather_codec_hooks(); -- 1.7.9.5

2 1

Re: [alsa-devel] ALSA: core: add support for compress_offload
by Dan Carpenter 05 Sep '12

05 Sep '12

Hello Vinod Koul, The patch b21c60a4edd2: "ALSA: core: add support for compress_offload" from Dec 23, 2011, leads to the following warning: sound/core/compress_offload.c:85 snd_compr_open() warn: bitwise AND condition is false here 83 if (f->f_flags & O_WRONLY) 84 dirn = SND_COMPRESS_PLAYBACK; 85 else if (f->f_flags & O_RDONLY) 86 dirn = SND_COMPRESS_CAPTURE; 87 else { 88 pr_err("invalid direction\n"); 89 return -EINVAL; 90 } O_RDONLY is 0 so we never do compressed capture. Perhaps the tests should be: if ((file->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; else if (file->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; else return -EINVAL; Also it seems like a bad thing to print an error here, because can the user flood dmesg? regards, dan carpenter

2 1

[alsa-devel] [patch] ASoC: wm0010: unlock on error path
by Dan Carpenter 05 Sep '12

05 Sep '12

We're holding the wm0010->lock mutex when we goto err_core. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This driver is only in linux-next. diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 5f99148..1c052d6 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -663,7 +663,9 @@ abort: wm0010_halt(codec); mutex_unlock(&wm0010->lock); return ret; + err_core: + mutex_unlock(&wm0010->lock); regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), wm0010->core_supplies); err:

2 1

Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Takashi Iwai 05 Sep '12

05 Sep '12

At Wed, 5 Sep 2012 15:00:15 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Applied this one, too. Thanks. Takashi > --- > sound/pci/emu10k1/memory.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c > index 0a43662..1dce0a39 100644 > --- a/sound/pci/emu10k1/memory.c > +++ b/sound/pci/emu10k1/memory.c > @@ -263,8 +263,8 @@ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *b > spin_lock_irqsave(&emu->memblk_lock, flags); > if (blk->mapped_page >= 0) { > /* update order link */ > - list_del(&blk->mapped_order_link); > - list_add_tail(&blk->mapped_order_link, &emu->mapped_order_link_head); > + list_move_tail(&blk->mapped_order_link, > + &emu->mapped_order_link_head); > spin_unlock_irqrestore(&emu->memblk_lock, flags); > return 0; > } > >

1 0

Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Takashi Iwai 05 Sep '12

05 Sep '12

At Wed, 5 Sep 2012 14:33:21 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied. Takashi > --- > sound/drivers/opl4/opl4_synth.c | 9 +++------ > 1 file changed, 3 insertions(+), 6 deletions(-) > > diff --git a/sound/drivers/opl4/opl4_synth.c b/sound/drivers/opl4/opl4_synth.c > index 49b9e24..4b91adc 100644 > --- a/sound/drivers/opl4/opl4_synth.c > +++ b/sound/drivers/opl4/opl4_synth.c > @@ -504,8 +504,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha > spin_lock_irqsave(&opl4->reg_lock, flags); > for (i = 0; i < voices; i++) { > voice[i] = snd_opl4_get_voice(opl4); > - list_del(&voice[i]->list); > - list_add_tail(&voice[i]->list, &opl4->on_voices); > + list_move_tail(&voice[i]->list, &opl4->on_voices); > voice[i]->chan = chan; > voice[i]->note = note; > voice[i]->velocity = vel & 0x7f; > @@ -555,8 +554,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha > > static void snd_opl4_voice_off(struct snd_opl4 *opl4, struct opl4_voice *voice) > { > - list_del(&voice->list); > - list_add_tail(&voice->list, &opl4->off_voices); > + list_move_tail(&voice->list, &opl4->off_voices); > > voice->reg_misc &= ~OPL4_KEY_ON_BIT; > snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); > @@ -571,8 +569,7 @@ void snd_opl4_note_off(void *private_data, int note, int vel, struct snd_midi_ch > > static void snd_opl4_terminate_voice(struct snd_opl4 *opl4, struct opl4_voice *voice) > { > - list_del(&voice->list); > - list_add_tail(&voice->list, &opl4->off_voices); > + list_move_tail(&voice->list, &opl4->off_voices); > > voice->reg_misc = (voice->reg_misc & ~OPL4_KEY_ON_BIT) | OPL4_DAMP_BIT; > snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); >

1 0

Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Clemens Ladisch 05 Sep '12

05 Sep '12

Wei Yongjun wrote: > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Acked-by: Clemens Ladisch <clemens(a)ladisch.de>

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[alsa-devel] [PATCH 0/3] HDMI HBR(High Bit Rate) Feature bug fix for Intel's Chips
by Wang Xingchao 05 Sep '12

05 Sep '12

Please Note This patchset has not been tested yet on any Hardware platform. i could only confirm there's no compile error. :) I test previous single patch on the Yamaha A/V receiver which works well, the test results was also confirmed by some guys from XBMC forum. The idea of this patch comes from Anssi, Big credit to him at first! Any comments are appreciated! Changelog: * do not touch channel_allocation priority order, use new api to get correct channels number * use spdif->status to check audio type, pcm or non-pcm * create new non_pcm_mapping[] for non_pcm audio in seperate block * add non_pcm hdmi_spec_per_cvt to remember previous audio type, used to determine whether need a channel mapping update during the pcm/non-pcm stream switch with same channel number. Wang Xingchao (3): ALSA: hdmi - fix channel_allocation array wrong order ALSA: HDMI - Enable HBR feature on Intel chips ALSA: HDMI - Setup channel mapping for non_pcm audio sound/pci/hda/patch_hdmi.c | 56 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 51 insertions(+), 5 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: HDMI - Enable HBR feature on Intel chips
by Wang Xingchao 05 Sep '12

05 Sep '12

HDMI channel remapping apparently effects HBR packets on Intel's chips. For compressed non-PCM audio, use "straight-through" channel mapping. For uncompressed multi-channel pcm audio, use normal mapping. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/patch_hdmi.c | 50 +++++++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d9439c5..fda89e3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -231,6 +231,11 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, /* Dolby Surround */ { .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, /* surround40 */ { .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, /* surround41 */ @@ -239,22 +244,15 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, /* surround51 */ { .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, { .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, { .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, { .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, @@ -537,22 +535,36 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, + hda_nid_t cvt_nid, int ca) { int i; int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); + unsigned short val; + int tmp_mapping[8]; + bool non_pcm = false; + + val = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + non_pcm = !!(val & AC_DIG1_NONAUDIO); + + if (hdmi_channel_mapping[ca][1] == 0 || non_pcm) { + for (i = 0; i < channel_allocations[ca].channels; i++) { + if (non_pcm) + tmp_mapping[i] = i | (i << 4); + else + hdmi_channel_mapping[ca][i] = i | (i << 4); + } for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + if (non_pcm) + tmp_mapping[i] = 0xf | (i << 4); + else + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); } for (i = 0; i < 8; i++) { err = snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); + non_pcm ? tmp_mapping[i] : hdmi_channel_mapping[ca][i]); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -686,7 +698,7 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, } static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - struct snd_pcm_substream *substream) + hda_nid_t cvt_nid, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; @@ -731,13 +743,13 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * sizeof(*dp_ai) to avoid partial match/update problems when * the user switches between HDMI/DP monitors. */ + hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, ca); if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, sizeof(ai))) { snd_printdd("hdmi_setup_audio_infoframe: " "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -1151,7 +1163,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, substream); + hdmi_setup_audio_infoframe(codec, pin_idx, cvt_nid, substream); pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- 1.7.9.5

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Re: [alsa-devel] How to define card specific pcm devices?
by Raymond Yau 05 Sep '12

05 Sep '12

2012-8-17 下午10:51 於 "Tanu Kaskinen" <tanu.kaskinen(a)digia.com> 寫道： > > Ping? > > If card specific pcm definitions are not possible, would it be > acceptable to have a generic "front_left:CARD" or "first_channel:CARD" > pcm, which would transform "front:CARD" or "hw:CARD" into a mono device? As pulseaudio server use SND_PCM_NO_AUTO_CHANNEL which diable channel conversion, this left channel device only work with other alsa applications except pulseaudio when your sound card does not support mono. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ff4af902cf4ac0… the solution is to change the mixing in pulseaudio by using left channel for this mic and those stereo mic with inverted phase , and average for mono playback device . stereo mic and line-in how about the 4 channel mic? > > -- Tanu > > > On Fri, 2012-08-10 at 14:25 +0300, Tanu Kaskinen wrote: > > Hi, > > > > I'm writing UCM configuration for RX-51 (Nokia N900), and I have a > > problem: "hw:RX51" can't be opened in mono mode, but there's only one > > integrated mono microphone, and its audio is routed to the left input > > channel. Applications will get only silence in the right channel when > > they use "hw:RX51". > > > > The best solution that I have come up is to define a pcm that converts > > "hw:RX51" into a mono device. The problem is that I don't know how > > exactly to do that. I guess the route plugin can be used for that: > > > > pcm.rx51_mono_from_left { > > type route > > slave.pcm hw:RX51 > > slave.channels 2 > > ttable.0.0 1 > > } > > > > I tried putting that definition in /usr/share/alsa/cards/RX-51.conf (and > > also RX51.conf, since it's not entirely clear to me when the dash should > > be omitted). I have this in the UCM configuration: > > > > Value { > > CapturePCM "rx51_mono_from_left" > > CaptureChannels "1" > > } > > > > This is printed to the PulseAudio log: > > > > (alsa-lib)pcm.c: Unknown PCM rx51_mono_from_left > > > > So the rx51_mono_from_left definition is not loaded. I'm not terribly > > surprised: I guess the card configuration files are loaded only when > > necessary, and when PulseAudio tries to open "rx51_mono_from_left", > > alsa-lib can't really know that it's somehow related to the RX-51 card. > > > > How should I solve this? > > > >

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[alsa-devel] [PATCH] ASoC: tegra: move platform data header
by Stephen Warren 05 Sep '12

05 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- .../sound/tegra_wm8903.h | 0 sound/soc/tegra/tegra_wm8903.c | 3 +-- 2 files changed, 1 insertions(+), 2 deletions(-) rename arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h => include/sound/tegra_wm8903.h (100%) diff --git a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h b/include/sound/tegra_wm8903.h similarity index 100% rename from arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h rename to include/sound/tegra_wm8903.h diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index d4f14e4..cee13b7 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,13 +34,12 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <mach/tegra_wm8903_pdata.h> - #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tegra_wm8903.h> #include "../codecs/wm8903.h" -- 1.7.0.4

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[alsa-devel] [RFC 0/3] Initial stab at converting OMAP ASoC support to DMA engine
by Russell King - ARM Linux 04 Sep '12

04 Sep '12

The following series of three patches is an attempt to convert the OMAP ASoC backend to use the DMA engine support. I'll bring your attention to the comments in patch 3 which highlight some of the features lost in this process. Some questions need answering there (in particular the one concerning pause/resume) especially as it seems the present driver could well be buggy wrt comments recently on the mailing list about ALSA pause/resume requirements. This works for me, given the wind is in the right direction and if it's sunny outside. I find the audio support on OMAP to be rather flakey at best, requiring reboots if it doesn't work first time.

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[alsa-devel] [PATCH 0/3] ALSA: snd-usb: Regression fixes for 3.5
by Daniel Mack 04 Sep '12

04 Sep '12

Here are 3 more regression fixes for snd-usb in 3.5. It's annoying they haven't been found earlier, but at least we're now back to the streaming quality we had before the streaming refactoring. Anyone with a USB device, please give it a quick shot at least. Thanks, Daniel Daniel Mack (3): ALSA: snd-usb: restore delay information ALSA: snd-usb: fix calls to next_packet_size ALSA: snd-usb: fix cross-interface streaming devices sound/usb/endpoint.c | 13 +------------ sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++---- 3 files changed, 49 insertions(+), 16 deletions(-) -- 1.7.11.4

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[alsa-devel] [PATCH v3 0/3] introduce device_buffer
by Vinod Koul 04 Sep '12

04 Sep '12

Old wine in a new bottle with added flavour now. Based on our discussions in LPC, now we will report the combined delay value as sum of the buffering done in driver and hardware. This is used to report delay Additionally we tell pcm what is buffer in device/driver, this is used to check for error calulcations Vinod Koul (3): ALSA: pcm - introduce device_buffer ASoC: add device_buffer in asoc ASoC: mid-x86 - implement buffer query in sst_platform driver include/sound/pcm.h | 1 + include/sound/soc-dai.h | 6 ++++++ include/sound/soc.h | 6 ++++++ sound/core/pcm_lib.c | 10 +++++++++- sound/soc/mid-x86/sst_platform.c | 23 +++++++++++++++++++++++ sound/soc/mid-x86/sst_platform.h | 4 +++- sound/soc/soc-pcm.c | 13 ++++++++++++- 7 files changed, 60 insertions(+), 3 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] ALSA: Add DSP firmware loader
by Ian Minett 04 Sep '12

04 Sep '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, Thanks for the recent code samples, they were very useful. We've built on them to produce this latest patchset, incorporating the updates you provided to add the loader functions, and doing a minimal update to the CA0132 codec to use the new ops to perform the firmware transfer. Please let us know if any tweaks are needed to the update. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's updates: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's updates: Pass DMA buffer pointers in calls to setup_bdle(). Add new callback routines to controller to handle loading firmware binaries. Add new switch to Kconfig to enable DSP firmware loader. 3: - patch_ca0132.c - ca0132_regs.h Add support for DSP firmware transfer to CA0132 codec: Add transfer handler functions. Call the new snd_hda_codec_load_dsp_xxx loader ops to control download. Add chip access routines.

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[alsa-devel] [PATCH] ALSA: snd-usb: Add quirks for Playback Designs devices
by Daniel Mack 04 Sep '12

04 Sep '12

Playback Designs' USB devices have some hardware limitations on their USB interface. In particular: - They need a 20ms delay after each class compliant request as the hardware ACKs the USB packets before the device is actually ready for the next command. Sending data immediately will result in buffer overflows in the hardware. - The devices send bogus feedback data at the start of each stream which confuse the feedback format auto-detection. This patch introduces a new quirks hook that is called after each control packet and which adds a delay for all devices that match Playback Designs' USB VID for now. In addition, it adds a counter to snd_usb_endpoint to drop received packets on the floor. Another new quirks function that is called once an endpoint is started initializes that counter for these devices on their sync endpoint. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Andreas Koch <andreas(a)akdesigninc.com> Supported-by: Demian Martin <demianm_1(a)yahoo.com> --- sound/usb/card.h | 2 ++ sound/usb/endpoint.c | 8 ++++++++ sound/usb/helper.c | 5 +++++ sound/usb/quirks.c | 24 ++++++++++++++++++++++++ sound/usb/quirks.h | 10 ++++++++++ 5 files changed, 49 insertions(+) diff --git a/sound/usb/card.h b/sound/usb/card.h index 2b9ffff..d9d2b5a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -92,6 +92,8 @@ struct snd_usb_endpoint { unsigned char silence_value; unsigned int stride; int iface, alt_idx; + int skip_packets; /* quirks for devices to ignore the first n packets + in a stream */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d6e2bb4..a83a18d 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -31,6 +31,7 @@ #include "card.h" #include "endpoint.h" #include "pcm.h" +#include "quirks.h" #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 @@ -170,6 +171,11 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep, { struct urb *urb = urb_ctx->urb; + if (unlikely(ep->skip_packets > 0)) { + ep->skip_packets--; + return; + } + if (ep->sync_slave) snd_usb_handle_sync_urb(ep->sync_slave, ep, urb); @@ -822,6 +828,8 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep) ep->unlink_mask = 0; ep->phase = 0; + snd_usb_endpoint_start_quirk(ep); + /* * If this endpoint has a data endpoint as implicit feedback source, * don't start the urbs here. Instead, mark them all as available, diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 9eed8f4..c1db28f 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -21,6 +21,7 @@ #include "usbaudio.h" #include "helper.h" +#include "quirks.h" /* * combine bytes and get an integer value @@ -97,6 +98,10 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, memcpy(data, buf, size); kfree(buf); } + + snd_usb_ctl_msg_quirk(dev, pipe, request, requesttype, + value, index, data, size); + return err; } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2781726..1eaacf1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -761,3 +761,27 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, } } +void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) +{ + /* + * "Playback Design" products send bogus feedback data at the start + * of the stream. Ignore them. + */ + if ((le16_to_cpu(ep->chip->dev->descriptor.idVendor) == 0x23ba) && + ep->type == SND_USB_ENDPOINT_TYPE_SYNC) + ep->skip_packets = 4; +} + +void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, + __u8 request, __u8 requesttype, __u16 value, + __u16 index, void *data, __u16 size) +{ + /* + * "Playback Design" products need a 20ms delay after each + * class compliant request + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(20); +} + diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 03e5e94..0ca9e91 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -1,6 +1,10 @@ #ifndef __USBAUDIO_QUIRKS_H #define __USBAUDIO_QUIRKS_H +struct audioformat; +struct snd_usb_endpoint; +struct snd_usb_substream; + int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, struct usb_driver *driver, @@ -20,4 +24,10 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp); +void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); + +void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, + __u8 request, __u8 requesttype, __u16 value, + __u16 index, void *data, __u16 size); + #endif /* __USBAUDIO_QUIRKS_H */ -- 1.7.11.4

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Re: [alsa-devel] [PATCH] ASoC: mc13783: Remove mono support
by Gaëtan Carlier 04 Sep '12

04 Sep '12

Hi, On 09/03/2012 06:04 PM, Fabio Estevam wrote: > Playing a mono track on a mc13783 codec results in incorrect playback rate. > > Remove mono support so that a mono track can be played correctly. > > Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> > --- > sound/soc/codecs/mc13783.c | 8 ++++---- > 1 file changed, 4 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c > index 8f726c0..115a403 100644 > --- a/sound/soc/codecs/mc13783.c > +++ b/sound/soc/codecs/mc13783.c > @@ -659,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { > .id = MC13783_ID_STEREO_DAC, > .playback = { > .stream_name = "Playback", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = MC13783_FORMATS, > @@ -670,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { > .id = MC13783_ID_STEREO_CODEC, > .capture = { > .stream_name = "Capture", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = MC13783_RATES_RECORD, > .formats = MC13783_FORMATS, > @@ -692,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { > .id = MC13783_ID_SYNC, > .playback = { > .stream_name = "Playback", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = MC13783_FORMATS, > }, > .capture = { > .stream_name = "Capture", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = MC13783_RATES_RECORD, > .formats = MC13783_FORMATS, > On imx27_3ds board: Tested-by: Gaëtan Carlier <gcembed(a)gmail.com> Regards, Gaëtan Carlier

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Re: [alsa-devel] [ASoC] Fix: Revert 'ASoC: imx-ssi: Remove mono support'.
by Gaëtan Carlier 04 Sep '12

04 Sep '12

Hi Javier, On 09/03/2012 10:27 AM, Javier Martin wrote: > The following commit should be reverted: 0865a75d4166bddc533fd50831829ceefb94f9b0 > > The bug this patch is meant to solve doesn't occur in Visstrim_M10 boards. > Furthermore, after applying this patch sound in Visstrim_M10 is played > at slower rates. > > Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> > --- > diff --git b/sound/soc/fsl/imx-ssi.c a/sound/soc/fsl/imx-ssi.c > index ac337ac..28dd76c 100644 > --- b/sound/soc/fsl/imx-ssi.c > +++ a/sound/soc/fsl/imx-ssi.c > @@ -380,13 +380,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) > static struct snd_soc_dai_driver imx_ssi_dai = { > .probe = imx_ssi_dai_probe, > .playback = { > - .channels_min = 2, > + .channels_min = 1, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = SNDRV_PCM_FMTBIT_S16_LE, > }, > .capture = { > - .channels_min = 2, > + .channels_min = 1, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = SNDRV_PCM_FMTBIT_S16_LE, > When applied on linux-next-20120824, the patch failed because original imx-ssi.c file looks like : static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &imx_ssi_pcm_dai_ops, }; The comment line is missing in your patch. Here is the right patch : @@ -378,18 +378,17 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) } static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &imx_ssi_pcm_dai_ops, Regards, Gaëtan Carlier.

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[alsa-devel] [Patch] Support for (original) Xbox Communicator
by Marko Friedemann 03 Sep '12

03 Sep '12

Hey guys, attached is a patch for the USB quirks to support the original Xbox Communicator (aka XboxLive Headset). I had actually submitted this some time ago against an earlier version but wasn't subscribed to the list and it was lost in moderation. I present it again now, against alsa 1.0.25. Tested and working under ubuntu 12.04.1 with kernel 3.2.0-29-generic. Regards, Marko -- "Somit werden Zitate zu zählbaren Einheiten wissenschaftlicher Reputation." -- NeMa-News - Trends in der Wissenschaft

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Re: [alsa-devel] [PATCH] sound: Remove the last mention of SNDRV_MAIN_OBJECT_FILE
by Takashi Iwai 03 Sep '12

03 Sep '12

At Sun, 2 Sep 2012 20:04:17 -0700, Josh Triplett wrote: > > SNDRV_MAIN_OBJECT_FILE hasn't done anything since the pre-git days, and > the only remaining reference occurs as a #define in sound/last.c. Drop > that last mention of it. > > Signed-off-by: Josh Triplett <josh(a)joshtriplett.org> Applied now. Thanks. Takashi > --- > sound/last.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/sound/last.c b/sound/last.c > index 7ffc182..43f2228 100644 > --- a/sound/last.c > +++ b/sound/last.c > @@ -19,7 +19,6 @@ > * > */ > > -#define SNDRV_MAIN_OBJECT_FILE > #include <linux/init.h> > #include <sound/core.h> > > -- > 1.7.10.4 >

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Re: [alsa-devel] [PATCH] ALSA: fix possible memory leak in snd_mixer_oss_build_input()
by Takashi Iwai 03 Sep '12

03 Sep '12

At Sun, 2 Sep 2012 22:10:27 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > uinfo has been allocated in this function and should be > freed before leaving from the error handling cases. > > spatch with a semantic match is used to found this problem. > (http://coccinelle.lip6.fr/) > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Applied now. Thanks. Takashi > --- > sound/core/oss/mixer_oss.c | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c > index 18297f7..29f6ded 100644 > --- a/sound/core/oss/mixer_oss.c > +++ b/sound/core/oss/mixer_oss.c > @@ -1046,6 +1046,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix > > if (kctl->info(kctl, uinfo)) { > up_read(&mixer->card->controls_rwsem); > + kfree(uinfo); > return 0; > } > strcpy(str, ptr->name); > @@ -1061,6 +1062,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix > uinfo->value.enumerated.item = slot.capture_item; > if (kctl->info(kctl, uinfo)) { > up_read(&mixer->card->controls_rwsem); > + kfree(uinfo); > return 0; > } > if (!strcmp(uinfo->value.enumerated.name, str)) { > >

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix runtime PM accounting
by Takashi Iwai 02 Sep '12

02 Sep '12

The flag codec->d3_clk_stop_ok isn't always set when the codec is powered off. If it's not set, the controller driver doesn't change the runtime PM count. This causes an unblanced accounting when multiple codecs are present. Usually, the first codec is powered down with CLK_STOP_OK=0, then the next codec with CLK_STOP_OK=1. At this moment, the first codec is changed also to CLK_STOP_OK=1, but this isn't informed to the controller. Thus it still thinks as if one codec is active. For fixing this issue, move the check of CLK_STOP_OK bit at the runtime suspend time, and simple up/down by the codec power save in azx_power_notifier(). For CLK_STOP_OK bit checks of all codecs, a new helper function is introduced. With this change, d3_clk_stop_ok field is no longer referred, thus removed. Also, d3_clk_stop bit field is moved to a place where more efficiently packed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I faced this issue while I'm checking the pm issue. Mengdong, could you verify whether this change is OK? sound/pci/hda/hda_codec.c | 30 ++++++++++++++++++++---------- sound/pci/hda/hda_codec.h | 6 +++--- sound/pci/hda/hda_intel.c | 4 ++-- 3 files changed, 25 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4a2f35c..b7ac1ff 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3546,10 +3546,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; -#ifdef CONFIG_SND_HDA_POWER_SAVE - codec->d3_stop_clk_ok = 0; -#endif - if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3572,12 +3568,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } - -#ifdef CONFIG_SND_HDA_POWER_SAVE - if ((power_state == AC_PWRST_D3) - && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) - codec->d3_stop_clk_ok = 1; -#endif } #ifdef CONFIG_SND_HDA_HWDEP @@ -4532,6 +4522,26 @@ void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) } EXPORT_SYMBOL_HDA(snd_hda_power_save); +/* check whether all codecs are ready to stop clock */ +bool snd_hda_bus_ready_for_d3(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + unsigned int state; + int fg = codec->afg ? codec->afg : codec->mfg; + + if (!codec->d3_stop_clk) + return false; + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (!(state & AC_PWRST_CLK_STOP_OK)) + return false; + } + return true; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_ready_for_d3); + /** * snd_hda_check_amp_list_power - Check the amp list and update the power * @codec: HD-audio codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 13c834f..abd85e5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -868,6 +868,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ @@ -875,9 +876,6 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; - - unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ @@ -1068,9 +1066,11 @@ const char *snd_hda_get_jack_location(u32 cfg); #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); +bool snd_hda_bus_ready_for_d3(struct hda_bus *bus); #else static inline void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) {} +static inline bool snd_hda_bus_ready_for_d3(struct hda_bus *bus) { return false; } #endif /** diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c9c779..9b02174 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2410,7 +2410,7 @@ static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + if (bus->power_keep_link_on) return; if (codec->power_on) @@ -2529,7 +2529,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; #ifdef CONFIG_SND_HDA_POWER_SAVE - if (!power_save_controller) + if (!power_save_controller || !snd_hda_bus_ready_for_d3(chip->bus)) return -EAGAIN; #endif -- 1.7.11.4

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[alsa-devel] [PATCH] ASoC: dapm: Ensure bypass paths are suspended and resumed
by Mark Brown 01 Sep '12

01 Sep '12

Since bypass paths aren't part of DAPM streams and we may not have any DAPM streams there may not be anything that triggers a DAPM sync for them. Mark all input and output widgets as dirty and then sync to do so at the end of suspend and resume. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 8 ++++++++ sound/soc/soc-dapm.c | 22 ++++++++++++++++++++++ 3 files changed, 31 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index abe373d..5b584cb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -412,6 +412,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 01ec2d6..3be2e40 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -609,6 +609,10 @@ int snd_soc_suspend(struct device *dev) SND_SOC_DAPM_STREAM_SUSPEND); } + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); + /* suspend all CODECs */ list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with @@ -756,6 +760,10 @@ static void soc_resume_deferred(struct work_struct *work) /* userspace can access us now we are back as we were before */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); + + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); } /* powers up audio subsystem after a suspend */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 53f69b5..fd97dd7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -141,6 +141,28 @@ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) } EXPORT_SYMBOL_GPL(dapm_mark_dirty); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_card *card = dapm->card; + struct snd_soc_dapm_widget *w; + + mutex_lock(&card->dapm_mutex); + + list_for_each_entry(w, &card->widgets, list) { + switch (w->id) { + case snd_soc_dapm_input: + case snd_soc_dapm_output: + dapm_mark_dirty(w, "Rechecking inputs and outputs"); + break; + default: + break; + } + } + + mutex_unlock(&card->dapm_mutex); +} +EXPORT_SYMBOL_GPL(dapm_mark_io_dirty); + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) -- 1.7.10.4

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Regression fixes for 3.5
by Daniel Mack 01 Sep '12

01 Sep '12

(please don't drop mailing lists from Cc:) On 31.08.2012 18:18, ☈king wrote: > On 08/31/2012 12:15 PM, Daniel Mack wrote: >> What kernel do you work on? Did you try the other patch that I sent >> some days ago and that already went mainline? >> https://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commit;h=01561… > > Hrm. > > I am using Gentoo's kernel from gentoo-sources. When I wrote the other > mail I was on 3.5.1, but now I'm using 3.5.3, and when I apply that > patch, I get: > > 1 out of 5 hunks FAILED -- saving rejects to file sound/usb/pcm.c.rej > % cat sound/usb/pcm.c.rej > --- sound/usb/pcm.c > +++ sound/usb/pcm.c > @@ -544,13 +544,10 @@ > subs->last_frame_number = 0; > runtime->delay = 0; > - /* clear the pending deactivation on the target EPs */ > - deactivate_endpoints(subs); > - > /* for playback, submit the URBs now; otherwise, the first hwptr_done > * updates for all URBs would happen at the same time when starting */ > if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) > - return start_endpoints(subs); > + return start_endpoints(subs, 1); > return 0; > } > > > It already doesn't have that deactivate_endpoints() line. > > So, I manually adding the 1 arg to start_endpoints() for "can_sleep". The important thing is not to call deactivate_endpoints() from the prepare callback. > Rebooting to this kernel gives the exact same results, though. Please try a kernel built from Takashi's branch: https://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git That alone should fix your problem. The three patches of this series fix some occasional stream dropouts. Thanks for testing, Daniel

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[alsa-devel] [PATCH] ASoC: samsung dma - Don't indicate support for pause/resume.
by Dylan Reid 01 Sep '12

01 Sep '12

The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..56a507a 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | -- 1.7.12.146.g16d26b1

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Re: [alsa-devel] no 5.1 surround sound on 3stack-6ch motherboard
by Raymond Yau 01 Sep '12

01 Sep '12

> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1044150 > > you need to switch channel mode from 2ch to 6ch and restore this > setting by alsactl before pulseaudio start > > Simple mixer control 'Channel Mode',0 > Capabilities: enum > Items: '2ch' '4ch' '6ch' > Item0: '2ch > > but how can i do this?? amixer -c0 set "Channel Mode" "6ch" alsactl store restart pulseaudio server http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=Do… Realtek codecs -------------- * Channel Mode This is an enum control to change the surround-channel setup, appears only when the surround channels are available. It gives the number of channels to be used, "2ch", "4ch", "6ch", and "8ch". According to the configuration, this also controls the jack-retasking of multi-I/O jacks. can you post the pulseaudio log ? I don't understand how can pulseaudio server switch to active profile surround51 when blue and pink jacks had not been retasked as output ( Pin-ctls are still IN ) and channel mode is still 2ch. analog-input rear mic and line-in ports should not be available too. http://www.intel.com/support/motherboards/desktop/sb/cs-020642.htm#6ch Jack retasking is a feature that allows an audio jack to support more than one function. For example, the pink audio jack can support microphone input when the system configuration is set to Stereo Mode, and as Center/LFE or Rear Surround (depending on the audio solution used) when set as 6 channel. Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19c50: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181345f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=05, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 index: 1 name: <alsa_card.pci-0000_00_1b.0> driver: <module-alsa-card.c> owner module: 5 properties: alsa.card = "0" alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0xf7d00000 irq 44" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" device.form_factor = "internal" device.string = "0" device.description = "Built-in Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" profiles: output:analog-stereo: Analogue Stereo Output (priority 1000) output:analog-stereo+input:analog-stereo: Analogue Stereo Duplex (priority 1010) output:analog-surround-40: Analogue Surround 4.0 Output (priority 700) output:analog-surround-40+input:analog-stereo: Analogue Surround 4.0 Output + Analogue Stereo Input (priority 710) output:analog-surround-41: Analogue Surround 4.1 Output (priority 800) output:analog-surround-41+input:analog-stereo: Analogue Surround 4.1 Output + Analogue Stereo Input (priority 810) output:analog-surround-50: Analogue Surround 5.0 Output (priority 700) output:analog-surround-50+input:analog-stereo: Analogue Surround 5.0 Output + Analogue Stereo Input (priority 710) output:analog-surround-51: Analogue Surround 5.1 Output (priority 3800) output:analog-surround-51+input:analog-stereo: Analogue Surround 5.1 Output + Analogue Stereo Input (priority 3810) input:analog-stereo: Analogue Stereo Input (priority 10) off: Off (priority 0) active profile: <output:analog-surround-51+input:analog-stereo> sinks: alsa_output.pci-0000_00_1b.0.analog-surround-51/#0: Built-in Audio Analogue Surround 5.1 sources: alsa_output.pci-0000_00_1b.0.analog-surround-51.monitor/#1: Monitor of Built-in Audio Analogue Surround 5.1 alsa_input.pci-0000_00_1b.0.analog-stereo/#2: Built-in Audio Analogue Stereo ports: analog-output: Analogue Output (priority 9900, available: unknown) properties: analog-output-headphones: Headphones (priority 9000, available: unknown) properties: analog-input-microphone-front: Front Microphone (priority 8500, available: no) properties: analog-input-microphone-rear: Rear Microphone (priority 8200, available: yes) properties: analog-input-linein: Line In (priority 8100, available: yes) properties:

1 0

[alsa-devel] [PATCH 6/9] ARM: OMAP: Move gpio.h to include/linux/platform_data
by Tony Lindgren 01 Sep '12

01 Sep '12

This way we can remove includes of plat/gpio.h which won't work with the single zImage support. Note that we also remove the cpu_class_is_omap2() check in gpio-omap.c as the drivers should not call it as we need to make it local to arch/arm/mach-omap2 for single zImage support. While at it, arrange the related includes in the standard way. Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: linux-mtd(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Acked-by: Tony Lindgren <tony(a)atomide.com> --- arch/arm/mach-omap1/board-ams-delta.c | 3 ++- arch/arm/mach-omap1/board-osk.c | 1 + arch/arm/mach-omap1/gpio15xx.c | 1 + arch/arm/mach-omap1/gpio16xx.c | 1 + arch/arm/mach-omap1/gpio7xx.c | 1 + arch/arm/mach-omap1/include/mach/gpio.h | 2 -- arch/arm/mach-omap1/leds-h2p2-debug.c | 4 +++- arch/arm/mach-omap1/leds.c | 1 + arch/arm/mach-omap2/board-am3517evm.c | 9 ++++++--- arch/arm/mach-omap2/board-cm-t35.c | 12 +++++++----- arch/arm/mach-omap2/board-zoom-display.c | 6 +++++- arch/arm/mach-omap2/board-zoom-peripherals.c | 1 + arch/arm/mach-omap2/gpio.c | 1 + arch/arm/mach-omap2/hsmmc.c | 5 ++++- arch/arm/mach-omap2/include/mach/gpio.h | 2 -- arch/arm/mach-omap2/msdi.c | 1 + arch/arm/mach-omap2/omap_hwmod_2420_data.c | 1 - arch/arm/mach-omap2/omap_hwmod_2430_data.c | 6 +++--- arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c | 2 +- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 6 +++--- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 6 +++--- arch/arm/mach-omap2/pm24xx.c | 1 + arch/arm/mach-omap2/pm34xx.c | 2 ++ drivers/gpio/gpio-omap.c | 15 ++++++++------- drivers/mtd/nand/ams-delta.c | 8 ++++++-- include/linux/platform_data/gpio-omap.h | 15 ++------------- sound/soc/omap/sdp3430.c | 1 + 27 files changed, 65 insertions(+), 49 deletions(-) rename arch/arm/plat-omap/include/plat/gpio.h => include/linux/platform_data/gpio-omap.h (94%) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 147d665..4b5578b 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -26,6 +26,8 @@ #include <linux/export.h> #include <linux/omapfb.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/keypad-omap.h> #include <media/soc_camera.h> @@ -35,7 +37,6 @@ #include <asm/mach/map.h> #include <plat/board-ams-delta.h> -#include <linux/platform_data/keypad-omap.h> #include <plat/mux.h> #include <mach/hardware.h> diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 9413fd3..c780a1b 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -39,6 +39,7 @@ #include <linux/mtd/partitions.h> #include <linux/mtd/physmap.h> #include <linux/i2c/tps65010.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/omap1_bl.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap1/gpio15xx.c b/arch/arm/mach-omap1/gpio15xx.c index ebef15e..98e6f39 100644 --- a/arch/arm/mach-omap1/gpio15xx.c +++ b/arch/arm/mach-omap1/gpio15xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1_MPUIO_VBASE OMAP1_MPUIO_BASE #define OMAP1510_GPIO_BASE 0xFFFCE000 diff --git a/arch/arm/mach-omap1/gpio16xx.c b/arch/arm/mach-omap1/gpio16xx.c index 2a48cd2..33f4192 100644 --- a/arch/arm/mach-omap1/gpio16xx.c +++ b/arch/arm/mach-omap1/gpio16xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1610_GPIO1_BASE 0xfffbe400 #define OMAP1610_GPIO2_BASE 0xfffbec00 diff --git a/arch/arm/mach-omap1/gpio7xx.c b/arch/arm/mach-omap1/gpio7xx.c index acf12b7..958ce9a 100644 --- a/arch/arm/mach-omap1/gpio7xx.c +++ b/arch/arm/mach-omap1/gpio7xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP7XX_GPIO1_BASE 0xfffbc000 #define OMAP7XX_GPIO2_BASE 0xfffbc800 diff --git a/arch/arm/mach-omap1/include/mach/gpio.h b/arch/arm/mach-omap1/include/mach/gpio.h index e737706..ebf86c0 100644 --- a/arch/arm/mach-omap1/include/mach/gpio.h +++ b/arch/arm/mach-omap1/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap1/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap1/leds-h2p2-debug.c b/arch/arm/mach-omap1/leds-h2p2-debug.c index f6b14a1..3ea12c7 100644 --- a/arch/arm/mach-omap1/leds-h2p2-debug.c +++ b/arch/arm/mach-omap1/leds-h2p2-debug.c @@ -14,13 +14,15 @@ #include <linux/kernel_stat.h> #include <linux/sched.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> #include <asm/leds.h> #include <asm/mach-types.h> #include <plat/fpga.h> +#include <mach/hardware.h> + #include "leds.h" diff --git a/arch/arm/mach-omap1/leds.c b/arch/arm/mach-omap1/leds.c index ae6dd93..7b1a383 100644 --- a/arch/arm/mach-omap1/leds.c +++ b/arch/arm/mach-omap1/leds.c @@ -6,6 +6,7 @@ #include <linux/gpio.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/leds.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap2/board-am3517evm.c b/arch/arm/mach-omap2/board-am3517evm.c index 00abda1..c6d63d7 100644 --- a/arch/arm/mach-omap2/board-am3517evm.c +++ b/arch/arm/mach-omap2/board-am3517evm.c @@ -25,19 +25,22 @@ #include <linux/can/platform/ti_hecc.h> #include <linux/davinci_emac.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> -#include <mach/am35xx.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> -#include "common.h" #include <plat/usb.h> + +#include <mach/hardware.h> +#include <mach/am35xx.h> + #include <video/omapdss.h> #include <video/omap-panel-generic-dpi.h> #include <video/omap-panel-tfp410.h> +#include "common.h" #include "am35xx-emac.h" #include "mux.h" #include "control.h" diff --git a/arch/arm/mach-omap2/board-cm-t35.c b/arch/arm/mach-omap2/board-cm-t35.c index ad83654..c2b2371 100644 --- a/arch/arm/mach-omap2/board-cm-t35.c +++ b/arch/arm/mach-omap2/board-cm-t35.c @@ -23,6 +23,9 @@ #include <linux/input/matrix_keypad.h> #include <linux/delay.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/mtd-nand-omap2.h> +#include <linux/platform_data/spi-omap2-mcspi.h> #include <linux/i2c/at24.h> #include <linux/i2c/twl.h> @@ -37,17 +40,16 @@ #include <asm/mach/arch.h> #include <asm/mach/map.h> -#include "common.h" -#include <linux/platform_data/mtd-nand-omap2.h> #include <plat/gpmc.h> #include <plat/usb.h> + +#include <mach/hardware.h> + #include <video/omapdss.h> #include <video/omap-panel-generic-dpi.h> #include <video/omap-panel-tfp410.h> -#include <linux/platform_data/spi-omap2-mcspi.h> - -#include <mach/hardware.h> +#include "common.h" #include "mux.h" #include "sdram-micron-mt46h32m32lf-6.h" #include "hsmmc.h" diff --git a/arch/arm/mach-omap2/board-zoom-display.c b/arch/arm/mach-omap2/board-zoom-display.c index 660a2932..0e48a51 100644 --- a/arch/arm/mach-omap2/board-zoom-display.c +++ b/arch/arm/mach-omap2/board-zoom-display.c @@ -15,9 +15,13 @@ #include <linux/i2c/twl.h> #include <linux/spi/spi.h> #include <linux/platform_data/spi-omap2-mcspi.h> -#include <video/omapdss.h> + +#include <plat/cpu.h> + #include <mach/board-zoom.h> +#include <video/omapdss.h> + #define LCD_PANEL_RESET_GPIO_PROD 96 #define LCD_PANEL_RESET_GPIO_PILOT 55 #define LCD_PANEL_QVGA_GPIO 56 diff --git a/arch/arm/mach-omap2/board-zoom-peripherals.c b/arch/arm/mach-omap2/board-zoom-peripherals.c index 00f73b7..ae5d36b 100644 --- a/arch/arm/mach-omap2/board-zoom-peripherals.c +++ b/arch/arm/mach-omap2/board-zoom-peripherals.c @@ -19,6 +19,7 @@ #include <linux/regulator/fixed.h> #include <linux/wl12xx.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> diff --git a/arch/arm/mach-omap2/gpio.c b/arch/arm/mach-omap2/gpio.c index 9ad7d48..7b52bdf 100644 --- a/arch/arm/mach-omap2/gpio.c +++ b/arch/arm/mach-omap2/gpio.c @@ -21,6 +21,7 @@ #include <linux/slab.h> #include <linux/interrupt.h> #include <linux/of.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/hsmmc.c b/arch/arm/mach-omap2/hsmmc.c index a9675d8..4b0e166 100644 --- a/arch/arm/mach-omap2/hsmmc.c +++ b/arch/arm/mach-omap2/hsmmc.c @@ -14,12 +14,15 @@ #include <linux/string.h> #include <linux/delay.h> #include <linux/gpio.h> -#include <mach/hardware.h> +#include <linux/platform_data/gpio-omap.h> + #include <plat/mmc.h> #include <plat/omap-pm.h> #include <plat/mux.h> #include <plat/omap_device.h> +#include <mach/hardware.h> + #include "mux.h" #include "hsmmc.h" #include "control.h" diff --git a/arch/arm/mach-omap2/include/mach/gpio.h b/arch/arm/mach-omap2/include/mach/gpio.h index be4d290..5621cc5 100644 --- a/arch/arm/mach-omap2/include/mach/gpio.h +++ b/arch/arm/mach-omap2/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap2/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap2/msdi.c b/arch/arm/mach-omap2/msdi.c index fb5bc6c..9e57b4a 100644 --- a/arch/arm/mach-omap2/msdi.c +++ b/arch/arm/mach-omap2/msdi.c @@ -23,6 +23,7 @@ #include <linux/kernel.h> #include <linux/err.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2420_data.c b/arch/arm/mach-omap2/omap_hwmod_2420_data.c index 558c300..b305d7f 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2420_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2420_data.c @@ -18,7 +18,6 @@ #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/dmtimer.h> #include <plat/l3_2xxx.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2430_data.c b/arch/arm/mach-omap2/omap_hwmod_2430_data.c index d662766..79ac236 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2430_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2430_data.c @@ -12,15 +12,15 @@ * XXX handle crossbar/shared link difference for L3? * XXX these should be marked initdata for multi-OMAP kernels */ +#include <linux/platform_data/dsp-mcbsp.h> +#include <linux/platform_data/spi-omap2-mcspi.h> + #include <plat/omap_hwmod.h> #include <mach/irqs.h> #include <plat/cpu.h> #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> -#include <linux/platform_data/dsp-mcbsp.h> -#include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/dmtimer.h> #include <plat/mmc.h> #include <plat/l3_2xxx.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c index 161ae0a..44e4764 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c @@ -10,9 +10,9 @@ */ #include <plat/omap_hwmod.h> #include <plat/serial.h> -#include <plat/gpio.h> #include <plat/dma.h> #include <plat/dmtimer.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/spi-omap2-mcspi.h> #include <mach/irqs.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c index 656bb7f..15f617e 100644 --- a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c @@ -15,6 +15,9 @@ * XXX these should be marked initdata for multi-OMAP kernels */ #include <linux/power/smartreflex.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/dsp-mcbsp.h> +#include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/omap_hwmod.h> #include <mach/irqs.h> @@ -24,10 +27,7 @@ #include <plat/l3_3xxx.h> #include <plat/l4_3xxx.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mmc.h> -#include <linux/platform_data/dsp-mcbsp.h> -#include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/dmtimer.h> #include "omap_hwmod_common_data.h" diff --git a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c index 3a06e7d..c9edaf2 100644 --- a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c @@ -19,15 +19,15 @@ */ #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/spi-omap2-mcspi.h> +#include <linux/platform_data/dsp-mcbsp.h> #include <linux/power/smartreflex.h> #include <plat/omap_hwmod.h> #include <plat/cpu.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/dma.h> -#include <linux/platform_data/spi-omap2-mcspi.h> -#include <linux/platform_data/dsp-mcbsp.h> #include <plat/mmc.h> #include <plat/dmtimer.h> #include <plat/common.h> diff --git a/arch/arm/mach-omap2/pm24xx.c b/arch/arm/mach-omap2/pm24xx.c index de1caf1..2708a8b 100644 --- a/arch/arm/mach-omap2/pm24xx.c +++ b/arch/arm/mach-omap2/pm24xx.c @@ -29,6 +29,7 @@ #include <linux/irq.h> #include <linux/time.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach/time.h> #include <asm/mach/irq.h> diff --git a/arch/arm/mach-omap2/pm34xx.c b/arch/arm/mach-omap2/pm34xx.c index e4fc88c..2a08978 100644 --- a/arch/arm/mach-omap2/pm34xx.c +++ b/arch/arm/mach-omap2/pm34xx.c @@ -28,6 +28,8 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/platform_data/gpio-omap.h> + #include <trace/events/power.h> #include <asm/suspend.h> diff --git a/drivers/gpio/gpio-omap.c b/drivers/gpio/gpio-omap.c index e6efd77..64fbce3 100644 --- a/drivers/gpio/gpio-omap.c +++ b/drivers/gpio/gpio-omap.c @@ -25,11 +25,9 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/irqdomain.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> -#include <asm/irq.h> -#include <mach/irqs.h> -#include <asm/gpio.h> #include <asm/mach/irq.h> #define OFF_MODE 1 @@ -385,13 +383,16 @@ static int _set_gpio_triggering(struct gpio_bank *bank, int gpio, static int gpio_irq_type(struct irq_data *d, unsigned type) { struct gpio_bank *bank = irq_data_get_irq_chip_data(d); - unsigned gpio; + unsigned gpio = 0; int retval; unsigned long flags; - if (!cpu_class_is_omap2() && d->irq > IH_MPUIO_BASE) +#ifdef CONFIG_ARCH_OMAP1 + if (d->irq > IH_MPUIO_BASE) gpio = OMAP_MPUIO(d->irq - IH_MPUIO_BASE); - else +#endif + + if (!gpio) gpio = irq_to_gpio(bank, d->irq); if (type & ~IRQ_TYPE_SENSE_MASK) diff --git a/drivers/mtd/nand/ams-delta.c b/drivers/mtd/nand/ams-delta.c index 861ca8f..78a524b 100644 --- a/drivers/mtd/nand/ams-delta.c +++ b/drivers/mtd/nand/ams-delta.c @@ -23,12 +23,16 @@ #include <linux/mtd/mtd.h> #include <linux/mtd/nand.h> #include <linux/mtd/partitions.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> + #include <asm/io.h> -#include <mach/hardware.h> #include <asm/sizes.h> -#include <linux/gpio.h> + #include <plat/board-ams-delta.h> +#include <mach/hardware.h> + /* * MTD structure for E3 (Delta) */ diff --git a/arch/arm/plat-omap/include/plat/gpio.h b/include/linux/platform_data/gpio-omap.h similarity index 94% rename from arch/arm/plat-omap/include/plat/gpio.h rename to include/linux/platform_data/gpio-omap.h index 50fb7cc..e8741c2 100644 --- a/arch/arm/plat-omap/include/plat/gpio.h +++ b/include/linux/platform_data/gpio-omap.h @@ -1,6 +1,4 @@ /* - * arch/arm/plat-omap/include/mach/gpio.h - * * OMAP GPIO handling defines and functions * * Copyright (C) 2003-2005 Nokia Corporation @@ -155,6 +153,8 @@ #define OMAP4_GPIO_CLEARDATAOUT 0x0190 #define OMAP4_GPIO_SETDATAOUT 0x0194 +#define OMAP_MAX_GPIO_LINES 192 + #define OMAP_MPUIO(nr) (OMAP_MAX_GPIO_LINES + (nr)) #define OMAP_GPIO_IS_MPUIO(nr) ((nr) >= OMAP_MAX_GPIO_LINES) @@ -213,16 +213,5 @@ extern void omap2_gpio_prepare_for_idle(int off_mode); extern void omap2_gpio_resume_after_idle(void); extern void omap_set_gpio_debounce(int gpio, int enable); extern void omap_set_gpio_debounce_time(int gpio, int enable); -/*-------------------------------------------------------------------------*/ - -/* - * Wrappers for "new style" GPIO calls, using the new infrastructure - * which lets us plug in FPGA, I2C, and other implementations. - * - * The original OMAP-specific calls should eventually be removed. - */ - -#include <linux/errno.h> -#include <asm-generic/gpio.h> #endif diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 33f4ac2..da45b51 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -33,6 +33,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/dsp-mcbsp.h> /* Register descriptions for twl4030 codec part */

2 1

Re: [alsa-devel] [PATCH] ASoC: correct the check for NULL dma_buffer pointer
by Mark Brown 01 Sep '12

01 Sep '12

On Fri, Aug 31, 2012 at 08:55:21AM +0530, Prasad Joshi wrote: > The if condition > if (!buf && !buf->area) > > checks if the buf pointer is NULL and then dereferences it again to > check if the buffer area is NULL, resulting in possible NULL > dereference. Applied, thanks.

1 0

[alsa-devel] [PATCH 0/3] introduce device_buffer
by Vinod Koul 01 Sep '12

01 Sep '12

Old wine ina new bottle, so here is the second attempt for adding the notion of how devices (in my case a DSP) can buffer data and that can be used by sound core Vinod Koul (3): ALSA: pcm - introduce device_buffer ASoC: add device_buffer in asoc ASoC: mid-x86 - implement buffer query in sst_platform driver include/sound/pcm.h | 1 + include/sound/soc-dai.h | 6 ++++++ include/sound/soc.h | 7 ++++++- sound/core/pcm_lib.c | 14 +++++++++++--- sound/core/pcm_native.c | 6 +++--- sound/soc/mid-x86/sst_platform.c | 23 +++++++++++++++++++++++ sound/soc/mid-x86/sst_platform.h | 4 +++- sound/soc/soc-pcm.c | 13 +++++++++++++ 8 files changed, 66 insertions(+), 8 deletions(-) -- 1.7.9.5

2 4

Re: [alsa-devel] [asoc:for-next 127/128] make[1]: *** No rule to make target `sound/soc/ep93xx'.
by Fengguang Wu 31 Aug '12

31 Aug '12

On Wed, Aug 29, 2012 at 08:13:38PM +0400, Alexander Shiyan wrote: > On Tue, 28 Aug 2012 22:38:56 -0700 > Fengguang Wu <fengguang.wu(a)intel.com> wrote: > > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > > head: 03a5da2b72ba426ed475795db375fc7780338ff4 > > commit: 0ed275eff31029c39355828cb48c46c0a006e2f8 [127/128] ASoC: Rename ep93xx soc directory to cirrus > > config: x86_64-allyesconfig (attached as .config) > > Build errors: > > make[1]: *** No rule to make target `sound/soc/ep93xx'. > > Cannot reproduce this bug. Possible this breakage may be resolved by run > "make distclean" before "make all". Sorry! It's actually a bug of the build system.. I just fixed it. Thanks, Fengguang

1 0

Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Regression fixes for 3.5
by ☈king 31 Aug '12

31 Aug '12

I am having the following symptoms, before and after application of this thread's patches: Given: - An onboard card - A Logitech h390 USB headset - alsa-info of: http://www.alsa-project.org/db/?f=942b5e96b53a667270f895ec5061ca14af0339ee When: - I do: aplay $(locate .wav | grep Front) it outputs: alsa ALSA lib pcm_direct.c:980:(snd1_pcm_direct_initialize_slave) unable to install hw params ALSA lib pcm_dmix.c:1030:(snd_pcm_dmix_open) unable to initialize slave aplay: main:682: audio open error: Broken pipe - I do a Skype "Make a test sound" /var/log/messages streams dozens of these: kernel: cannot submit urb 0, error -28: not enough bandwidth Note that this is a new laptop and headset, so I haven't compared how either works on other setups, such as with other kernels or alsa versions. Any hints? Any further info needed? Thanks! —☈

2 2

[alsa-devel] double dai question and analog paths
by Raffaele Recalcati 31 Aug '12

31 Aug '12

I can play succesfully each of the following substreams: static struct snd_soc_dai_link omap3baia_dai[] = { { .name = "TLV320AIC31", .stream_name = "AIC31", .cpu_dai_name = "omap-mcbsp-dai.3", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", .init = omap3baia_tlvaic31_init, .ops = &omap3baia_tlvaic31_ops, }, { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai_name = "omap-mcbsp-dai.1", .platform_name = "omap-pcm-audio", .codec_dai_name = "twl4030-hifi", .codec_name = "twl4030-codec", .init = omap3baia_twl4030_init, .ops = &omap3baia_twl4030_ops, }, }; I have also some analog connections between TWL4030 and TLV320AIC31 and I'd like to create an automatic audio path (for example from TWL4030 HFL to TLV320AIC31 LINE2L), but I obtain "Failed to add route Virtual Tps to Tlv->LINE2L" static const struct snd_soc_dapm_route audio_map_twl4030[] = { {"Virtual Tps to Tlv", NULL, "HFL"}, /* from tps to tlv */ {"LINE2L", NULL, "Virtual Tps to Tlv"}, /* from tps to tlv */ }; Because LINE2L is obviously defined inside TWL4030 codec. static const struct snd_soc_dapm_widget dapm_widgets_twl4030[] = { SND_SOC_DAPM_OUTPUT("Virtual Tps to Tlv"), }; Is there a clean way to manage this situation? I have also tried without "Virtual Tps to Tlv" but the problem is the same. Thx, Raffaele

1 0

[alsa-devel] [PATCH] ALSA: snd-usb: use list_for_each_safe for endpoint resources
by Pavel Roskin 31 Aug '12

31 Aug '12

snd_usb_endpoint_free() frees the structure that contains its argument. Signed-off-by: Pavel Roskin <proski(a)gnu.org> Cc: stable(a)vger.kernel.org --- sound/usb/card.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d5b5c33..4a469f0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, struct snd_usb_audio *chip) { struct snd_card *card; - struct list_head *p; + struct list_head *p, *n; if (chip == (void *)-1L) return; @@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_stream_disconnect(p); } /* release the endpoint resources */ - list_for_each(p, &chip->ep_list) { + list_for_each_safe(p, n, &chip->ep_list) { snd_usb_endpoint_free(p); } /* release the midi resources */

2 1

Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 30 Aug '12

30 Aug '12

On 25.08.2012 14:17, Josh Boyer wrote: > On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: >> On 25.08.2012 14:07, Bruno Wolff III wrote: >>> On Sat, Aug 25, 2012 at 14:02:51 +0200, >>> Daniel Mack <zonque(a)gmail.com> wrote: >>>> >>>> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >>>> scheduling-while-atomic bug in PCM capture stream") and see if that >>> >>> I can try that, but it takes a long time to build a new kernel on my >>> old hardware. >>> >>>> helps? If not, can you summarize again which kernels still work for you >>>> and which don't? >>> >>> The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The earliest that >>> doesn't work is 3.6.0-0.rc2.git2.1.fc18. >>> >> >> The report you sent doesn't look like it could be caused by e9ba389c5. >> It fixes a kernel Ooops. But as it is the only relevant patch in that >> area, it would be interesting if reverting it fixes anything. > > Yep, agreed. If this revert kernel doesn't work, we're likely down to a > git bisect, Bruno. > >> Btw - thanks a lot for testing -rc kernels, much appreciated! > > Indeed! Could you please try this patch on top of Takashi's? Thanks again! Daniel

2 6

[alsa-devel] [PATCH 04/22] ASoC: Ux500: Move MSP pinctrl setup into the MSP driver
by Lee Jones 30 Aug '12

30 Aug '12

In the initial submission of the MSP driver msp1 and msp3's associated pinctrl mechanism was passed back to platform code using a plat_init() call-back routine, but it has no place in platform code. The MSP driver should set this up for the appropriate ports. Instead we use a use_pinctrl identifier which is passed from platform_data/Device Tree which indicates which ports should use pinctrl. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- arch/arm/mach-ux500/board-mop500-msp.c | 75 +------------------------------- arch/arm/mach-ux500/include/mach/msp.h | 2 - sound/soc/ux500/ux500_msp_i2s.c | 75 +++++++++++++++++++++++++------- sound/soc/ux500/ux500_msp_i2s.h | 2 - 4 files changed, 60 insertions(+), 94 deletions(-) diff --git a/arch/arm/mach-ux500/board-mop500-msp.c b/arch/arm/mach-ux500/board-mop500-msp.c index df15646..fc78d5d 100644 --- a/arch/arm/mach-ux500/board-mop500-msp.c +++ b/arch/arm/mach-ux500/board-mop500-msp.c @@ -23,53 +23,6 @@ #include "devices-db8500.h" #include "pins-db8500.h" -/* MSP1/3 Tx/Rx usage protection */ -static DEFINE_SPINLOCK(msp_rxtx_lock); - -/* Reference Count */ -static int msp_rxtx_ref; - -/* Pin modes */ -struct pinctrl *msp1_p; -struct pinctrl_state *msp1_def; -struct pinctrl_state *msp1_sleep; - -int msp13_i2s_init(void) -{ - int retval = 0; - unsigned long flags; - - spin_lock_irqsave(&msp_rxtx_lock, flags); - if (msp_rxtx_ref == 0 && !(IS_ERR(msp1_p) || IS_ERR(msp1_def))) { - retval = pinctrl_select_state(msp1_p, msp1_def); - if (retval) - pr_err("could not set MSP1 defstate\n"); - } - if (!retval) - msp_rxtx_ref++; - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - - return retval; -} - -int msp13_i2s_exit(void) -{ - int retval = 0; - unsigned long flags; - - spin_lock_irqsave(&msp_rxtx_lock, flags); - WARN_ON(!msp_rxtx_ref); - msp_rxtx_ref--; - if (msp_rxtx_ref == 0 && !(IS_ERR(msp1_p) || IS_ERR(msp1_sleep))) { - retval = pinctrl_select_state(msp1_p, msp1_sleep); - if (retval) - pr_err("could not set MSP1 sleepstate\n"); - } - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - - return retval; -} - static struct stedma40_chan_cfg msp0_dma_rx = { .high_priority = true, .dir = STEDMA40_PERIPH_TO_MEM, @@ -132,8 +85,6 @@ static struct msp_i2s_platform_data msp1_platform_data = { .id = MSP_I2S_1, .msp_i2s_dma_rx = NULL, .msp_i2s_dma_tx = &msp1_dma_tx, - .msp_i2s_init = msp13_i2s_init, - .msp_i2s_exit = msp13_i2s_exit, }; static struct stedma40_chan_cfg msp2_dma_rx = { @@ -219,47 +170,23 @@ static struct msp_i2s_platform_data msp3_platform_data = { .id = MSP_I2S_3, .msp_i2s_dma_rx = &msp1_dma_rx, .msp_i2s_dma_tx = NULL, - .msp_i2s_init = msp13_i2s_init, - .msp_i2s_exit = msp13_i2s_exit, }; int mop500_msp_init(struct device *parent) { - struct platform_device *msp1; - pr_info("%s: Register platform-device 'snd-soc-mop500'.\n", __func__); platform_device_register(&snd_soc_mop500); pr_info("Initialize MSP I2S-devices.\n"); db8500_add_msp_i2s(parent, 0, U8500_MSP0_BASE, IRQ_DB8500_MSP0, &msp0_platform_data); - msp1 = db8500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, + db8500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, &msp1_platform_data); db8500_add_msp_i2s(parent, 2, U8500_MSP2_BASE, IRQ_DB8500_MSP2, &msp2_platform_data); db8500_add_msp_i2s(parent, 3, U8500_MSP3_BASE, IRQ_DB8500_MSP1, &msp3_platform_data); - /* Get the pinctrl handle for MSP1 */ - if (msp1) { - msp1_p = pinctrl_get(&msp1->dev); - if (IS_ERR(msp1_p)) - dev_err(&msp1->dev, "could not get MSP1 pinctrl\n"); - else { - msp1_def = pinctrl_lookup_state(msp1_p, - PINCTRL_STATE_DEFAULT); - if (IS_ERR(msp1_def)) { - dev_err(&msp1->dev, - "could not get MSP1 defstate\n"); - } - msp1_sleep = pinctrl_lookup_state(msp1_p, - PINCTRL_STATE_SLEEP); - if (IS_ERR(msp1_sleep)) - dev_err(&msp1->dev, - "could not get MSP1 idlestate\n"); - } - } - pr_info("%s: Register platform-device 'ux500-pcm'\n", __func__); platform_device_register(&ux500_pcm); diff --git a/arch/arm/mach-ux500/include/mach/msp.h b/arch/arm/mach-ux500/include/mach/msp.h index 798be19..3cc7142 100644 --- a/arch/arm/mach-ux500/include/mach/msp.h +++ b/arch/arm/mach-ux500/include/mach/msp.h @@ -22,8 +22,6 @@ struct msp_i2s_platform_data { enum msp_i2s_id id; struct stedma40_chan_cfg *msp_i2s_dma_rx; struct stedma40_chan_cfg *msp_i2s_dma_tx; - int (*msp_i2s_init) (void); - int (*msp_i2s_exit) (void); }; #endif diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 36be11e..2cbfc54 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/pinctrl/consumer.h> #include <linux/delay.h> #include <linux/slab.h> @@ -25,6 +26,17 @@ #include "ux500_msp_i2s.h" +/* MSP1/3 Tx/Rx usage protection */ +static DEFINE_SPINLOCK(msp_rxtx_lock); + +/* Pin modes */ +struct pinctrl *pinctrl_p; +struct pinctrl_state *pinctrl_def; +struct pinctrl_state *pinctrl_sleep; + +/* Reference Count */ +int pinctrl_rxtx_ref; + /* Protocol desciptors */ static const struct msp_protdesc prot_descs[] = { { /* I2S */ @@ -352,17 +364,23 @@ static int configure_multichannel(struct ux500_msp *msp, static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) { - int status = 0; + int status = 0, retval = 0; u32 reg_val_DMACR, reg_val_GCR; + unsigned long flags; /* Check msp state whether in RUN or CONFIGURED Mode */ - if ((msp->msp_state == MSP_STATE_IDLE) && (msp->plat_init)) { - status = msp->plat_init(); - if (status) { - dev_err(msp->dev, "%s: ERROR: Failed to init MSP (%d)!\n", - __func__, status); - return status; + if (msp->msp_state == MSP_STATE_IDLE) { + spin_lock_irqsave(&msp_rxtx_lock, flags); + if (pinctrl_rxtx_ref == 0 && + !(IS_ERR(pinctrl_p) || IS_ERR(pinctrl_def))) { + retval = pinctrl_select_state(pinctrl_p, + pinctrl_def); + if (retval) + pr_err("could not set MSP defstate\n"); } + if (!retval) + pinctrl_rxtx_ref++; + spin_unlock_irqrestore(&msp_rxtx_lock, flags); } /* Configure msp with protocol dependent settings */ @@ -620,7 +638,8 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) { - int status = 0; + int status = 0, retval = 0; + unsigned long flags; dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir); @@ -631,12 +650,19 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) writel((readl(msp->registers + MSP_GCR) & (~(FRAME_GEN_ENABLE | SRG_ENABLE))), msp->registers + MSP_GCR); - if (msp->plat_exit) - status = msp->plat_exit(); - if (status) - dev_warn(msp->dev, - "%s: WARN: ux500_msp_i2s_exit failed (%d)!\n", - __func__, status); + + spin_lock_irqsave(&msp_rxtx_lock, flags); + WARN_ON(!pinctrl_rxtx_ref); + pinctrl_rxtx_ref--; + if (pinctrl_rxtx_ref == 0 && + !(IS_ERR(pinctrl_p) || IS_ERR(pinctrl_sleep))) { + retval = pinctrl_select_state(pinctrl_p, + pinctrl_sleep); + if (retval) + pr_err("could not set MSP sleepstate\n"); + } + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + writel(0, msp->registers + MSP_GCR); writel(0, msp->registers + MSP_TCF); writel(0, msp->registers + MSP_RCF); @@ -678,8 +704,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->id = platform_data->id; msp->dev = &pdev->dev; - msp->plat_init = platform_data->msp_i2s_init; - msp->plat_exit = platform_data->msp_i2s_exit; msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; @@ -717,6 +741,25 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); msp->i2s_cont = i2s_cont; + pinctrl_p = pinctrl_get(msp->dev); + if (IS_ERR(pinctrl_p)) + dev_err(&pdev->dev, "could not get MSP pinctrl\n"); + else { + pinctrl_def = pinctrl_lookup_state(pinctrl_p, + PINCTRL_STATE_DEFAULT); + if (IS_ERR(pinctrl_def)) { + dev_err(&pdev->dev, + "could not get MSP defstate (%li)\n", + PTR_ERR(pinctrl_def)); + } + pinctrl_sleep = pinctrl_lookup_state(pinctrl_p, + PINCTRL_STATE_SLEEP); + if (IS_ERR(pinctrl_sleep)) + dev_err(&pdev->dev, + "could not get MSP idlestate (%li)\n", + PTR_ERR(pinctrl_def)); + } + return 0; err_i2s_cont: diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 2d9136d..1ce336f 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -524,8 +524,6 @@ struct ux500_msp { struct dma_chan *rx_pipeid; enum msp_state msp_state; int (*transfer) (struct ux500_msp *msp, struct i2s_message *message); - int (*plat_init) (void); - int (*plat_exit) (void); struct timer_list notify_timer; int def_elem_len; unsigned int dir_busy; -- 1.7.9.5

2 5

Re: [alsa-devel] [PATCH v2 02/14] MFD: twl4030-audio: Convert to use devm_kzalloc
by Peter Ujfalusi 30 Aug '12

30 Aug '12

Hi Tero, On 08/28/2012 12:50 PM, Tero Kristo wrote: > On Tue, 2012-08-14 at 17:22 +0300, Peter Ujfalusi wrote: >> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > > I think this one could use a short commit message, also about why > kfree():s are dropped (handled internally by devm_* etc.) I don't like empty commit messages either but this patch is self explaining I would think that the commit title explains it. -- Péter

1 0

Re: [alsa-devel] REGRESSION: Webcam mic for 145f:015b not longer working with 1.0.24 and above (OK with 1.0.23)
by Daniel Mack 30 Aug '12

30 Aug '12

On 27.08.2012 12:33, Kostas wrote: > On 27/08/2012 12:26 μμ, Daniel Mack wrote: >> On 27.08.2012 11:00, Kostas wrote: >>> Hi All, >>> >>> I have a 'Trust Webcam' (USB ID: 145f:015b) that comes with a microphone. >>> The microphone works with Alsa 1.0.23 but it does not work with the >>> latest Alsa 1.0.25 neither with Alsa 1.0.24. >>> Apparently this is a regression. >>> >>> What shall I do to figure out the source of the problem? >>> >>> Here are attached my alsa-infos, with Alsa 1.0.23 (working), 1.0.24 (NOT >>> working) and 1.0.25 (NOT working). >> Can you provide dmesg protocols for both a working and a non-working >> case please? >> >> > Below are attached the dmesg files. > Thanks - the logs look both good, I guess the interesting messages are only generated once you try to start the stream. Can you try this?

2 22

[alsa-devel] NRPACKS setting
by Raymond Celius 29 Aug '12

29 Aug '12

Hi, Is the NRPACKS setting equal to the number of packets per URB ? Best regards, Raymond Den 29. aug.. 2012 kl. 16.14 skrev alsa-devel-request(a)alsa-project.org: > Send Alsa-devel mailing list submissions to > alsa-devel(a)alsa-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > or, via email, send a message with subject or body 'help' to > alsa-devel-request(a)alsa-project.org > > You can reach the person managing the list at > alsa-devel-owner(a)alsa-project.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Alsa-devel digest..." > > > Today's Topics: > > 1. Re: [PATCH] ALSA: hda - Fix runtime PM accounting (Lin, Mengdong) > 2. Re: Logitech USB headset not working in 3.6-rc3 (Takashi Iwai) > 3. Re: Logitech USB headset not working in 3.6-rc3 (Daniel Mack) > 4. Re: [PATCH] ALSA: hda - Fix runtime PM accounting (Takashi Iwai) > 5. Re: Logitech USB headset not working in 3.6-rc3 (Takashi Iwai) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 29 Aug 2012 12:19:32 +0000 > From: "Lin, Mengdong" <mengdong.lin(a)intel.com> > To: Takashi Iwai <tiwai(a)suse.de> > Cc: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> > Subject: Re: [alsa-devel] [PATCH] ALSA: hda - Fix runtime PM > accounting > Message-ID: > <F46914AEC2663F4A9BB62374E5EEF8F81DAFDF(a)SHSMSX101.ccr.corp.intel.com> > Content-Type: text/plain; charset="us-ascii" > >> Now I introduced codec->epss flag that is initialized once in >> snd_hda_codec_new(), and let the codec driver overrides this flag. >> One bonus is that you can avoid the unnecessary codec parameter >> read at >> each power change. > > Yes, this is better. And the patch works well for me - [PATCH 1/2] > ALSA: hda - Avoid unnecessary parameter read for EPSS. > >> In anyway, there are a few issues: >> - The refcounting gets broken after module unload. It's because we >> leave the codec's pm_runtime usages after removal. > > Thanks for pointing out this! But I think the following codec can > introduce risk of race among multiple codecs to modify ' chip- > >power_count', > because only one codec's power on/off is serialized. > static void azx_power_notify(struct hda_bus *bus, struct hda_codec > *codec) > { > ... > if (codec->power_on) { > pm_runtime_get_sync(&chip->pci->dev); > chip->power_count++; > } else { > pm_runtime_put_sync(&chip->pci->dev); > chip->power_count--; > } > } > > How about calling pm_runtime_put_noidle() when freeing a codec in D0? > Maybe we can add a bus notify function like this: > Static void azx_codec_free_notify(struct hda_codec *codec) > { > If(codec->power_on) > pm_runtime_put_noidle(&codec->bus->pci->dev); > } > And snd_hda_codec_free() can trigger this notify. > >> - The D3-stop-clock check isn't done properly for the codecs >> providing >> own set_power_state ops. > > I cannot test the branch 'if (codec->patch_ops.set_power_state)' in > hda_set_power_state() for lack of such a codec. > But the logic seems right and the other branch works well for me :-) > >> - Ditto for the repeated power-set sequence, the error check, etc. > In hda_sync_power_state(), > if ((state & 0xff) == power_state) ... should be if ((state & > 0xf0)>> 4 == power_state), because PS-Act is bits 7:4 > break; > > And would you please explain why changing to D3 need not wait? One > of my codec does have a delay on change from D0 to D3. > > Thanks > Mengdong > > > > ------------------------------ > > Message: 2 > Date: Wed, 29 Aug 2012 15:29:03 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > To: Daniel Mack <zonque(a)gmail.com> > Cc: Bruno Wolff III <bruno(a)wolff.to>, alsa-devel(a)alsa-project.org, > linux-kernel(a)vger.kernel.org, Josh Boyer <jwboyer(a)redhat.com> > Subject: Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3 > Message-ID: <s5hsjb56d40.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=US-ASCII > > At Wed, 29 Aug 2012 13:26:25 +0200, > Daniel Mack wrote: >> >> [1 <text/plain; ISO-8859-1 (7bit)>] >> On 25.08.2012 14:17, Josh Boyer wrote: >>> On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: >>>> On 25.08.2012 14:07, Bruno Wolff III wrote: >>>>> On Sat, Aug 25, 2012 at 14:02:51 +0200, >>>>> Daniel Mack <zonque(a)gmail.com> wrote: >>>>>> >>>>>> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >>>>>> scheduling-while-atomic bug in PCM capture stream") and see if >>>>>> that >>>>> >>>>> I can try that, but it takes a long time to build a new kernel >>>>> on my >>>>> old hardware. >>>>> >>>>>> helps? If not, can you summarize again which kernels still work >>>>>> for you >>>>>> and which don't? >>>>> >>>>> The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The >>>>> earliest that >>>>> doesn't work is 3.6.0-0.rc2.git2.1.fc18. >>>>> >>>> >>>> The report you sent doesn't look like it could be caused by >>>> e9ba389c5. >>>> It fixes a kernel Ooops. But as it is the only relevant patch in >>>> that >>>> area, it would be interesting if reverting it fixes anything. >>> >>> Yep, agreed. If this revert kernel doesn't work, we're likely >>> down to a >>> git bisect, Bruno. >>> >>>> Btw - thanks a lot for testing -rc kernels, much appreciated! >>> >>> Indeed! >> >> Could you please try this patch on top of Takashi's? Thanks again! >> >> >> Daniel >> >> [2 0001-ALSA-snd-usb-Fix-URB-cancellation-at-stream-start.patch >> <text/x-patch (7bit)>] >>> From 6617bb2463ae3fad21390b59cc2a71f96b9e9ca8 Mon Sep 17 00:00:00 >>> 2001 >> From: Daniel Mack <zonque(a)gmail.com> >> Date: Wed, 29 Aug 2012 13:17:05 +0200 >> Subject: [PATCH] ALSA: snd-usb: Fix URB cancellation at stream start >> >> Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug >> in >> PCM capture stream") fixed a scheduling-while-atomic bug that >> happened >> when snd_usb_endpoint_start was called from the trigger callback, >> which >> is an atmic context. However, the patch breaks the idea of the >> endpoints >> reference counting, which is the reason why the driver has been >> refactored lately. >> >> Revert that commit and let snd_usb_endpoint_start() take care of >> the URB >> cancellation again. As this function is called from both atomic and >> non-atomic context, add a flag to denote whether the function may >> sleep. >> >> Signed-off-by: Daniel Mack <zonque(a)gmail.com> >> Cc: stable(a)kernel.org [3.5+] >> --- >> sound/usb/endpoint.c | 10 ++++++++-- >> sound/usb/endpoint.h | 2 +- >> sound/usb/pcm.c | 13 +++++-------- >> 3 files changed, 14 insertions(+), 11 deletions(-) >> >> diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c >> index c411812..678456c 100644 >> --- a/sound/usb/endpoint.c >> +++ b/sound/usb/endpoint.c >> @@ -799,7 +799,9 @@ int snd_usb_endpoint_set_params(struct >> snd_usb_endpoint *ep, >> /** >> * snd_usb_endpoint_start: start an snd_usb_endpoint >> * >> - * @ep: the endpoint to start >> + * @ep: the endpoint to start >> + * @can_sleep: flag indicating whether the operation is executed in >> + * non-atomic context >> * >> * A call to this function will increment the use count of the >> endpoint. >> * In case it is not already running, the URBs for this endpoint >> will be >> @@ -809,7 +811,7 @@ int snd_usb_endpoint_set_params(struct >> snd_usb_endpoint *ep, >> * >> * Returns an error if the URB submission failed, 0 in all other >> cases. >> */ >> -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) >> +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int >> can_sleep) >> { >> int err; >> unsigned int i; >> @@ -821,6 +823,10 @@ int snd_usb_endpoint_start(struct >> snd_usb_endpoint *ep) >> if (++ep->use_count != 1) >> return 0; >> >> + /* just to be sure */ >> + deactivate_urbs(ep, 0, can_sleep); >> + wait_clear_urbs(ep); > > It'd be safer to protect the call of wait_clear_urbs() when > can_sleep=0. > > > thanks, > > Takashi > > > ------------------------------ > > Message: 3 > Date: Wed, 29 Aug 2012 15:32:34 +0200 > From: Daniel Mack <zonque(a)gmail.com> > To: Takashi Iwai <tiwai(a)suse.de> > Cc: Bruno Wolff III <bruno(a)wolff.to>, alsa-devel(a)alsa-project.org, > linux-kernel(a)vger.kernel.org, Josh Boyer <jwboyer(a)redhat.com> > Subject: Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3 > Message-ID: <503E19F2.3080302(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 29.08.2012 15:29, Takashi Iwai wrote: >> At Wed, 29 Aug 2012 13:26:25 +0200, >> Daniel Mack wrote: >>> >>> [1 <text/plain; ISO-8859-1 (7bit)>] >>> On 25.08.2012 14:17, Josh Boyer wrote: >>>> On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: >>>>> On 25.08.2012 14:07, Bruno Wolff III wrote: >>>>>> On Sat, Aug 25, 2012 at 14:02:51 +0200, >>>>>> Daniel Mack <zonque(a)gmail.com> wrote: >>>>>>> >>>>>>> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >>>>>>> scheduling-while-atomic bug in PCM capture stream") and see if >>>>>>> that >>>>>> >>>>>> I can try that, but it takes a long time to build a new kernel >>>>>> on my >>>>>> old hardware. >>>>>> >>>>>>> helps? If not, can you summarize again which kernels still >>>>>>> work for you >>>>>>> and which don't? >>>>>> >>>>>> The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The >>>>>> earliest that >>>>>> doesn't work is 3.6.0-0.rc2.git2.1.fc18. >>>>>> >>>>> >>>>> The report you sent doesn't look like it could be caused by >>>>> e9ba389c5. >>>>> It fixes a kernel Ooops. But as it is the only relevant patch in >>>>> that >>>>> area, it would be interesting if reverting it fixes anything. >>>> >>>> Yep, agreed. If this revert kernel doesn't work, we're likely >>>> down to a >>>> git bisect, Bruno. >>>> >>>>> Btw - thanks a lot for testing -rc kernels, much appreciated! >>>> >>>> Indeed! >>> >>> Could you please try this patch on top of Takashi's? Thanks again! >>> >>> >>> Daniel >>> >>> [2 0001-ALSA-snd-usb-Fix-URB-cancellation-at-stream-start.patch >>> <text/x-patch (7bit)>] >>>> From 6617bb2463ae3fad21390b59cc2a71f96b9e9ca8 Mon Sep 17 00:00:00 >>>> 2001 >>> From: Daniel Mack <zonque(a)gmail.com> >>> Date: Wed, 29 Aug 2012 13:17:05 +0200 >>> Subject: [PATCH] ALSA: snd-usb: Fix URB cancellation at stream start >>> >>> Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic >>> bug in >>> PCM capture stream") fixed a scheduling-while-atomic bug that >>> happened >>> when snd_usb_endpoint_start was called from the trigger callback, >>> which >>> is an atmic context. However, the patch breaks the idea of the >>> endpoints >>> reference counting, which is the reason why the driver has been >>> refactored lately. >>> >>> Revert that commit and let snd_usb_endpoint_start() take care of >>> the URB >>> cancellation again. As this function is called from both atomic and >>> non-atomic context, add a flag to denote whether the function may >>> sleep. >>> >>> Signed-off-by: Daniel Mack <zonque(a)gmail.com> >>> Cc: stable(a)kernel.org [3.5+] >>> --- >>> sound/usb/endpoint.c | 10 ++++++++-- >>> sound/usb/endpoint.h | 2 +- >>> sound/usb/pcm.c | 13 +++++-------- >>> 3 files changed, 14 insertions(+), 11 deletions(-) >>> >>> diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c >>> index c411812..678456c 100644 >>> --- a/sound/usb/endpoint.c >>> +++ b/sound/usb/endpoint.c >>> @@ -799,7 +799,9 @@ int snd_usb_endpoint_set_params(struct >>> snd_usb_endpoint *ep, >>> /** >>> * snd_usb_endpoint_start: start an snd_usb_endpoint >>> * >>> - * @ep: the endpoint to start >>> + * @ep: the endpoint to start >>> + * @can_sleep: flag indicating whether the operation is executed in >>> + * non-atomic context >>> * >>> * A call to this function will increment the use count of the >>> endpoint. >>> * In case it is not already running, the URBs for this endpoint >>> will be >>> @@ -809,7 +811,7 @@ int snd_usb_endpoint_set_params(struct >>> snd_usb_endpoint *ep, >>> * >>> * Returns an error if the URB submission failed, 0 in all other >>> cases. >>> */ >>> -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) >>> +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int >>> can_sleep) >>> { >>> int err; >>> unsigned int i; >>> @@ -821,6 +823,10 @@ int snd_usb_endpoint_start(struct >>> snd_usb_endpoint *ep) >>> if (++ep->use_count != 1) >>> return 0; >>> >>> + /* just to be sure */ >>> + deactivate_urbs(ep, 0, can_sleep); >>> + wait_clear_urbs(ep); >> >> It'd be safer to protect the call of wait_clear_urbs() when >> can_sleep=0. > > Right. New patch attached. > >

1 0

[alsa-devel] [PATCH V2 0/6] ARM/ASoC: Davinci: Add Device Tree Support
by Hebbar, Gururaja 29 Aug '12

29 Aug '12

This patch-set is a cumulative of below tasks 1. Split asp header into platform and audio specific Davinci driver is shared by few OMAP platforms (like TI81xx, AM335x). Splitting asp header into Davinci platform specific and Audio specific helps to share them across platforms. 2. Add DT support for Davinci Audio Module (McASP, machine, pcm & Codec) This patch was tested on AM18x Board (Tested to check if the commits doesn't break on compile or runtime) and is based on repo at http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git; a=shortlog;h=refs/heads/for-3.7 (v3.6-rc2) Note: Matt Porter from TI is working on DMA engine conversion of DaVinci EDMA support and DT conversion of DMA parts will be taken up post that is complete. Since Audio requires DMA to work, this patch set does not yet provide complete functionality required to use sound on DT-based platforms, but is a step towards that direction. Changes from V1: - Instanciate seudo-device (davinci-pcm) directly from DAI driver rather than having a separate virtual-device registered - add more explanation to DT parameters - Change DT parameter from "ti,codec-clock" to "ti,codec-clock-rate" - use standard DT GPIO reference by phandle to populate gpio based DT parameters - Dont return err if some DT parameters are missing. They will be treated as 0. - Through Error warning if num-serializer != "serial-dir" array size - Coding Style correction Hebbar, Gururaja (6): ASoC: davinci: davinci-pcm does not need to be a plaform_driver ARM: Davinci: Remove references to davinci pcm ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific ASoC: Davinci: McASP: add device tree support for McASP ASoC: Davinci: machine: Add device tree binding ASoC: tlv320aic3x: Add device tree bindings .../bindings/sound/davinci-evm-audio.txt | 63 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 44 +++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 20 ++ arch/arm/mach-davinci/asp.h | 49 +++++ arch/arm/mach-davinci/davinci.h | 3 +- arch/arm/mach-davinci/devices-da8xx.c | 8 +- arch/arm/mach-davinci/devices.c | 11 - arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- .../linux/platform_data/davinci_asp.h | 73 ++----- sound/soc/codecs/tlv320aic3x.c | 31 +++ sound/soc/davinci/davinci-evm.c | 203 +++++++++++++++++--- sound/soc/davinci/davinci-i2s.c | 13 +- sound/soc/davinci/davinci-mcasp.c | 134 +++++++++++++- sound/soc/davinci/davinci-mcasp.h | 3 +- sound/soc/davinci/davinci-pcm.c | 24 +-- sound/soc/davinci/davinci-pcm.h | 6 +- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/davinci/davinci-vcif.c | 8 + 22 files changed, 573 insertions(+), 132 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt create mode 100644 arch/arm/mach-davinci/asp.h rename arch/arm/mach-davinci/include/mach/asp.h => include/linux/platform_data/davinci_asp.h (56%)

3 16

Re: [alsa-devel] [PATCHv2] sound: fix emu8000 DRAM sized 512 KiB too small
by Takashi Iwai 28 Aug '12

28 Aug '12

At Mon, 27 Aug 2012 22:25:21 -0400, David Flater wrote: > > v2: Fixed result still wrong in the case of 512 KiB DRAM. Oops. > > Applicable to 3.5.3 mainline. > > In emu8000.c, size_dram determines the amount of memory on the sound card by > doing write/readback tests starting at 512 KiB and incrementing by 512 KiB. > On success, detected_size is updated to the successful address and testing > continues. On failure, the loop is immediately exited. The resulting > detected_size is 512 KiB too small except in two special cases: > > 1. If there is no memory, the initial 0 value of detected_size is used, which > is correct. > 2. If the address space wraps around, detected_size is updated before the > bailout, so the result is correct. > > The patch corrects all cases and was tested with an AWE64 Gold. Before: > EMU8000 [0x620]: 3584 Kb on-board memory detected > asfxload 4GMGSMT.SF2 (4174814 B) fails. > After: > EMU8000 [0x620]: 4096 Kb on-board memory detected > asfxload 4GMGSMT.SF2 succeeds. > > I do not have a card with 512 KiB to test with, but by forcibly enabling the > added conditional I verified on the AWE64 Gold that it detects 512 KiB > (successfully reading from the first memory location) and does not hang the > card. > > C.f. Bug 46451 https://bugzilla.kernel.org/show_bug.cgi?id=46451 > > Signed-off-by: David Flater <dave(a)flaterco.com> Thanks, applied. Takashi > --- > > --- linux-3.5.3-nameless/sound/isa/sb/emu8000.c.orig 2012-08-25 22:32:13.000000000 -0400 > +++ linux-3.5.3-nameless/sound/isa/sb/emu8000.c 2012-08-27 22:06:15.729510037 -0400 > @@ -417,9 +417,6 @@ size_dram(struct snd_emu8000 *emu) > EMU8000_SMLD_READ(emu); /* discard stale data */ > if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) > break; /* no memory at this address */ > - > - detected_size = size; > - > snd_emu8000_read_wait(emu); > > /* > @@ -432,6 +429,18 @@ size_dram(struct snd_emu8000 *emu) > if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) > break; /* we must have wrapped around */ > snd_emu8000_read_wait(emu); > + > + /* Otherwise, it's valid memory. */ > + detected_size = size + 512 * 1024; > + } > + > + /* Distinguish 512 KiB from 0. */ > + if (detected_size == 0) { > + snd_emu8000_read_wait(emu); > + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); > + EMU8000_SMLD_READ(emu); /* discard stale data */ > + if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) > + detected_size = 512 * 1024; > } > > /* wait until FULL bit in SMAxW register is false */ > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ > -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

Re: [alsa-devel] [PATCH] ASoC: Rename ep93xx soc directory to cirrus
by Mark Brown 28 Aug '12

28 Aug '12

On Tue, Aug 21, 2012 at 07:21:51PM +0400, Alexander Shiyan wrote: > This patch is to rename the directory "ep93xx" in "cirrus". > Name more accurately reflects the manufacturer and allows to add > drivers not only for architecture ep93xx in this directory. > Patch not contain any functional changes. Applied, thanks.

1 0

[alsa-devel] [RESUBMIT PATCH] ASoC: SAMSUNG: Change Kconfig to support all SAMSUNG ASoC
by Sangsu Park 28 Aug '12

28 Aug '12

All SAMSUNG ASoC needs SND_SOC_SAMSUNG configuration. This patch change Kconfig to support all SAMSUNG ASoC. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f17dd25..e7b8317 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on PLAT_SAMSUNG select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help -- 1.7.1 On 08/28/12 8:00, Kukjin Kim wrote: > On 08/27/12 11:08, Mark Brown wrote: > > On Mon, Aug 27, 2012 at 11:06:47AM -0700, Kukjin Kim wrote: > > > >> Hmm, all of Samsung stuff is available...so how about? > >> > >> + depends on PLAT_SAMSUNG > > > > Yes, that's a very good idea! > > Mark, thanks. > > Sangsu, could you please re-submit as per my suggestion? > > Thanks. > Kukjin, thanks. This is re-submit patch. In fact, I'd wanted to submit your idea :) Thanks. > Best regards, > Kgene. > -- > Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung > Electronics Co., Ltd.

3 2

[alsa-devel] [PATCH] sound: tegra_alc5632: remove HP detect GPIO inversion
by Stephen Warren 28 Aug '12

28 Aug '12

From: Stephen Warren <swarren(a)nvidia.com> Both the schematics and practical testing show that the HP detect GPIO is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio should not specify to invert the signal. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Cc: Marc Dietrich <marvin24(a)gmx.de> Cc: Leon Romanovsky <leon(a)leon.nu> Cc: <stable(a)vger.kernel.org> # v3.4 v3.5 --- sound/soc/tegra/tegra_alc5632.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e463529..76cb1b3 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { .name = "Headset detection", .report = SND_JACK_HEADSET, .debounce_time = 150, - .invert = 1, }; static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { -- 1.7.9.5

4 3

Re: [alsa-devel] [PATCH v2 02/14] MFD: twl4030-audio: Convert to use devm_kzalloc
by Mark Brown 28 Aug '12

28 Aug '12

On Tue, Aug 28, 2012 at 12:50:10PM +0300, Tero Kristo wrote: > On Tue, 2012-08-14 at 17:22 +0300, Peter Ujfalusi wrote: > > Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > I think this one could use a short commit message, also about why > kfree():s are dropped (handled internally by devm_* etc.) Honestly I think people should be able to figure this out, it's not like it's an obscure API nobody ever used before.

1 0

[alsa-devel] [PATCH] ALSA: hda: Make core jack usage unconditional
by Mark Brown 28 Aug '12

28 Aug '12

Currently the HDA drivers have a Kconfig variable SND_HDA_INPUT_JACK which disables jack reporting via the ALSA core jack API. This is bad since (modulo the current problems with the HDA-specific reporting API) it means that all reporting mechanisms will get disabled, not just input. It's also not good that this is open coded in an individual driver - if it's useful to have this control it's going to be equally useful for any driver with jack detection support so should be handled in the core. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/pci/hda/Kconfig | 9 +-------- sound/pci/hda/hda_jack.c | 8 -------- 2 files changed, 1 insertion(+), 16 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 194d625..2c7f626 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -2,6 +2,7 @@ menuconfig SND_HDA_INTEL tristate "Intel HD Audio" select SND_PCM select SND_VMASTER + select SND_JACK if INPUT=y || INPUT=SND select SND_KCTL_JACK help Say Y here to include support for Intel "High Definition @@ -62,14 +63,6 @@ config SND_HDA_INPUT_BEEP_MODE Set 1 to always enable the digital beep interface for HD-audio by default. -config SND_HDA_INPUT_JACK - bool "Support jack plugging notification via input layer" - depends on INPUT=y || INPUT=SND - select SND_JACK - help - Say Y here to enable the jack plugging notification via - input layer. - config SND_HDA_PATCH_LOADER bool "Support initialization patch loading for HD-audio" depends on EXPERIMENTAL diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index aaccc02..0760d75 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -109,7 +109,6 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_new); void snd_hda_jack_tbl_clear(struct hda_codec *codec) { -#ifdef CONFIG_SND_HDA_INPUT_JACK /* free jack instances manually when clearing/reconfiguring */ if (!codec->bus->shutdown && codec->jacktbl.list) { struct hda_jack_tbl *jack = codec->jacktbl.list; @@ -119,7 +118,6 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) snd_device_free(codec->bus->card, jack->jack); } } -#endif snd_array_free(&codec->jacktbl); } @@ -225,16 +223,13 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) continue; state = get_jack_plug_state(jack->pin_sense); snd_kctl_jack_report(codec->bus->card, jack->kctl, state); -#ifdef CONFIG_SND_HDA_INPUT_JACK if (jack->jack) snd_jack_report(jack->jack, state ? jack->type : 0); -#endif } } EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); -#ifdef CONFIG_SND_HDA_INPUT_JACK /* guess the jack type from the pin-config */ static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { @@ -261,7 +256,6 @@ static void hda_free_jack_priv(struct snd_jack *jack) jacks->nid = 0; jacks->jack = NULL; } -#endif /** * snd_hda_jack_add_kctl - Add a kctl for the given pin @@ -292,7 +286,6 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); -#ifdef CONFIG_SND_HDA_INPUT_JACK if (!phantom_jack) { jack->type = get_input_jack_type(codec, nid); err = snd_jack_new(codec->bus->card, name, jack->type, @@ -303,7 +296,6 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->jack->private_free = hda_free_jack_priv; snd_jack_report(jack->jack, state ? jack->type : 0); } -#endif return 0; } -- 1.7.10.4

2 2

[alsa-devel] ASoC: SAMSUNG: Change Kconfig to support all EXYNOSs
by 박상수 28 Aug '12

28 Aug '12

All EXYNOSs use SND_SOC_SAMSUNG configuration. This patch change Kconfig to support all EXYNOSs like EXYNOS4 and EXYNOS5. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f17dd25..9e64ae6 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help -- 1.7.1

3 4

[alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Ricardo Neri 27 Aug '12

27 Aug '12

Hello! I have been working on prototypes for the ASoC OMAP HDMI audio driver to propagate events from the HDMI output (e.g., display getting enabled/disabled/suspended). This for the users of the driver to react to such events. For instance, if the display is disabled or disconected, audio could be stopped, rerouted or whatever other decision the user makes. This is needed because, if, for instance, the HDMI IP goes off, audio will stall and the audio users will only see a "playback write error (DMA or IRQ trouble?)" In my prototypes I have used snd_soc_jack for this purpose and I have some questions: *I see snd_soc_jack is used mostly for headsets and microphones with actual external mechanical connections. Strictly, in my case I propagate events originated by the OMAP display driver (changes in the power state), and not from external events. Some of these events are generated from an actual HDMI cable connection/disconnection, though. *Maybe the event should be propagated by omapdss/omapdrm/drm and the entity in charge of the audio policy should listen those events instead. *I do see SND_JACK_VIDEOOUT and SND_JACK_AVOUT types so maybe it is feasible for an audio driver to report events from an AV output. I was wondering about how much sense does it make to you guys use a snd_soc_jack in this case? Thanks in advance for your comments! Ricardo

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[alsa-devel] Kernel module build workflow (Re: Logitech USB headset not working in 3.6-rc3)
by Takashi Iwai 27 Aug '12

27 Aug '12

At Sat, 25 Aug 2012 07:16:21 -0500, Bruno Wolff III wrote: > > On Sat, Aug 25, 2012 at 08:13:27 -0400, > Josh Boyer <jwboyer(a)redhat.com> wrote: > >On Sat, Aug 25, 2012 at 07:07:40AM -0500, Bruno Wolff III wrote: > >> On Sat, Aug 25, 2012 at 14:02:51 +0200, > >> Daniel Mack <zonque(a)gmail.com> wrote: > >> > > >> >Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix > >> >scheduling-while-atomic bug in PCM capture stream") and see if that > >> > >> I can try that, but it takes a long time to build a new kernel on my > >> old hardware. > > > >I started a scratch build of a kernel with that patch reverted (via > >patch -R). You'll find it here when it finishes: > > > >http://koji.fedoraproject.org/koji/taskinfo?taskID=4421817 > > > >josh > > I'll test it when it finishes. Thanks. It takes me a long time to build > kernels locally, and I wasn't sure how to do a scratch build on koji without > committing stuff remotely. Actually you don't have to rebuild the whole kernel at all just for debugging this usb-audio bug. The workflow below is what I do often for debugging a driver problem on a distro kernel. 1. Prepare the compile-ready kernel source tree corresponding to your running kernel. This may depend on distro; in the case of SUSE, kernel-default-devel.rpm must be installed for kernel-default.rpm, for example. 2. Copy sound/usb subdirectory of the kernel source tree locally anywhere else, say, $HOME/usb. It can be a normal user directory. 3. Apply the patch to that local directory. 4. Build modules as a normal user like % make -C /lib/modules/$(uname -r)/build M=$HOME/usb modules 5. Make an extra update directory in the module directory once, e.g. # mkdir /lib/modules/$(uname -r)/updates 6. Copy *.ko files to there # cp $HOME/usb/*.ko /lib/modules/$(uname -r)/updates 7. Run depmod once # /sbin/depmod -a 8. Reload snd-usb-audio module. If you want to take back to the original module, just remove updates module directory and run depmod again. Pretty convenient, eh? HTH, Takashi

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Re: [alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Use module_spi_driver
by Mark Brown 27 Aug '12

27 Aug '12

On Mon, Aug 27, 2012 at 05:00:26PM +0530, Sachin Kamat wrote: > module_spi_driver makes the code simpler by eliminating > module_init and module_exit calls. Applied, thanks.

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Re: [alsa-devel] [PATCH - pcxhr 1/1] ALSA: pcxhr: Add 8 new sound cards
by Takashi Iwai 27 Aug '12

27 Aug '12

At Mon, 27 Aug 2012 09:03:55 +0200, Markus Bollinger wrote: > > Hi Takashi, > > sorry, I had problems to get git-send-email working, so I tried it on > another way, but it seems that was not the right one.... > Please find the patch attached. Generated from alsa.kmirror.git. Thanks, applied now. Takashi

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Re: [alsa-devel] [pulseaudio-discuss] Documentation for how rewinding works
by Raymond Yau 27 Aug '12

27 Aug '12

2012-8-19 下午7:17 於 "Tanu Kaskinen" <tanuk(a)iki.fi> 寫道： > > Hi, > > While reviewing a rewind related patch, I noticed that I don't > understand the rewinding code thoroughly, even though I have studied it > a few times in the past. I decided that I should write a wiki page for > the code so that working with it would be easier in the future for me > and everybody else. > > So, here's the result: > http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer… > > While doing that, I found quite a few bugs, which are noted on that > page. I don't plan fixing them any time soon (help with that would be > very welcome), so I will just file bugs about the issues so that they > are not forgotten. > it seem that the alsa sink is more complicated than your figure. Refer to figure 14 HD Audio DMA and buffering , section 4.6 Energy Efficient HA audio mechanism most sound cards have fixed size FIFO buffer instead of dynamic FIFO and minimum transfer size(brust) similar to hda controller which snd_pcm_rewind cannot rewind any audio data in those first in first out buffer. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa?i… 1) do the server really need to rewind when there is only one pulse client ? 2) does pulseaudio add this rewind safeguard delay to the latency ?

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[alsa-devel] REGRESSION: Webcam mic for 145f:015b not longer working with 1.0.24 and above (OK with 1.0.23)
by Kostas 27 Aug '12

27 Aug '12

Hi All, I have a 'Trust Webcam' (USB ID: 145f:015b) that comes with a microphone. The microphone works with Alsa 1.0.23 but it does not work with the latest Alsa 1.0.25 neither with Alsa 1.0.24. Apparently this is a regression. What shall I do to figure out the source of the problem? Here are attached my alsa-infos, with Alsa 1.0.23 (working), 1.0.24 (NOT working) and 1.0.25 (NOT working). Thanks! 1.0.23 http://www.alsa-project.org/db/?f=a4e2c86d93ff67c5d6acd4933c76f47a60cf51ee 1.0.24 http://www.alsa-project.org/db/?f=9aa13f70ec169011b9dfc5625370ce3e6f7827a1 1.0.25 http://www.alsa-project.org/db/?f=c3f63b69655b379e3eec54ab00e63b186a677422

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[alsa-devel] Driver for HT|OMEGA eCLARO (C-Media 8788 ***PCIe***)
by Aaron Fulkerson 27 Aug '12

27 Aug '12

Hello and nice to meet you all. This will be my first post if the previous one did not work. The problem is this: there is not a working driver for the HT|OMEGA eCLARO. The card is based on the C-Media 8788 chip for which there is an existing driver, however, due to the fact that the eCLARO is a PCIe-based card, the driver doesn't function properly. What are the chances of a working driver being written for this card?

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[alsa-devel] [PATCH v2 0/4] ASoC: omap-mcbsp: Sidetone related changes
by Peter Ujfalusi 25 Aug '12

25 Aug '12

Hello, Changes since v1: - Macro for the ST control has been fixed - New patch to combine the st channel volume set/get macros Mark: this series depends on the McBSP DT support (to be precise patch 3 and 4 has dependency). Intro mail from v1: When we boot with DT blob the sidetone is not supported at the moment. Prepare the McBSP code for the time when we are going to be able to support ST with DT booted kernel. Meanwhile do not block the audio card if the st_data is missing from the port, but print a warning about it. This will allow us to debug the DT booted kernels with audio. Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-mcbsp: Check mcbsp->id instead of cpu_dai->id when adding ST controls ASoC: omap-mcbsp: Only print warning if the st_data is missing for the port ASoC: omap-mcbsp: Use macro to create the McBSP2/3 ST controls ASoC: omap-mcbsp: Single macro for st channel volume set/get sound/soc/omap/omap-mcbsp.c | 64 ++++++++++++++++++------------------------- 1 files changed, 27 insertions(+), 37 deletions(-) -- 1.7.8.6

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[alsa-devel] [PATCH] ASoC: wm0010: Move resource acquisition to device probe
by Mark Brown 25 Aug '12

25 Aug '12

This is more idimatic for modern drivers. Also fix a couple of return codes while we're at it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 54 ++++++++++++++++++++++----------------------- 1 file changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 30ec0bd..5f99148 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -799,14 +799,35 @@ static irqreturn_t wm0010_irq(int irq, void *data) static int wm0010_probe(struct snd_soc_codec *codec) { struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); - struct spi_device *spi = to_spi_device(wm0010->dev); + + wm0010->codec = codec; + + return 0; +} + +static int __devinit wm0010_spi_probe(struct spi_device *spi) +{ unsigned long flags; unsigned long gpio_flags; int ret; int trigger; int irq; + struct wm0010_priv *wm0010; - wm0010->codec = codec; + wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), + GFP_KERNEL); + if (!wm0010) + return -ENOMEM; + + mutex_init(&wm0010->lock); + spin_lock_init(&wm0010->irq_lock); + + spi_set_drvdata(spi, wm0010); + wm0010->dev = &spi->dev; + + if (dev_get_platdata(&spi->dev)) + memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), + sizeof(wm0010->pdata)); init_completion(&wm0010->boot_completion); @@ -850,7 +871,7 @@ static int wm0010_probe(struct snd_soc_codec *codec) } } else { dev_err(wm0010->dev, "No reset GPIO configured\n"); - return ret; + return -EINVAL; } irq = spi->irq; @@ -862,9 +883,11 @@ static int wm0010_probe(struct snd_soc_codec *codec) ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, "wm0010", wm0010); - if (ret) + if (ret) { dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", irq, ret); + return ret; + } wm0010->irq = irq; if (spi->max_speed_hz) @@ -876,29 +899,6 @@ static int wm0010_probe(struct snd_soc_codec *codec) wm0010->state = WM0010_POWER_OFF; spin_unlock_irqrestore(&wm0010->irq_lock, flags); - return 0; -} - -static int __devinit wm0010_spi_probe(struct spi_device *spi) -{ - struct wm0010_priv *wm0010; - int ret; - - wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), - GFP_KERNEL); - if (!wm0010) - return -ENOMEM; - - mutex_init(&wm0010->lock); - spin_lock_init(&wm0010->irq_lock); - - spi_set_drvdata(spi, wm0010); - wm0010->dev = &spi->dev; - - if (dev_get_platdata(&spi->dev)) - memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), - sizeof(wm0010->pdata)); - ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm0010, wm0010_dai, ARRAY_SIZE(wm0010_dai)); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: wm0010: Add dummy widget for CLKIN
by Mark Brown 25 Aug '12

25 Aug '12

Make it easier to integrate the management of the clock supplying the WM0010 with DAPM by providing a dummy supply widget which supplies the interface widgets, this can be connected to clock outputs by the machines. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f820b57..b6f7097 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -118,9 +118,18 @@ struct wm0010_spi_msg { size_t len; }; +static const struct snd_soc_dapm_widget wm0010_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("CLKIN", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { { "SDI2 Capture", NULL, "SDI1 Playback" }, { "SDI1 Capture", NULL, "SDI2 Playback" }, + + { "SDI1 Capture", NULL, "CLKIN" }, + { "SDI2 Capture", NULL, "CLKIN" }, + { "SDI1 Playback", NULL, "CLKIN" }, + { "SDI2 Playback", NULL, "CLKIN" }, }; static const char *wm0010_state_to_str(enum wm0010_state state) @@ -718,6 +727,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .set_bias_level = wm0010_set_bias_level, .set_sysclk = wm0010_set_sysclk, + .dapm_widgets = wm0010_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm0010_dapm_widgets), .dapm_routes = wm0010_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), }; -- 1.7.10.4

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Takashi Iwai 25 Aug '12

25 Aug '12

At Sat, 25 Aug 2012 08:17:58 -0400, Josh Boyer wrote: > > On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: > > On 25.08.2012 14:07, Bruno Wolff III wrote: > > > On Sat, Aug 25, 2012 at 14:02:51 +0200, > > > Daniel Mack <zonque(a)gmail.com> wrote: > > >> > > >> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix > > >> scheduling-while-atomic bug in PCM capture stream") and see if that > > > > > > I can try that, but it takes a long time to build a new kernel on my > > > old hardware. > > > > > >> helps? If not, can you summarize again which kernels still work for you > > >> and which don't? > > > > > > The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The earliest that > > > doesn't work is 3.6.0-0.rc2.git2.1.fc18. > > > > > > > The report you sent doesn't look like it could be caused by e9ba389c5. > > It fixes a kernel Ooops. But as it is the only relevant patch in that > > area, it would be interesting if reverting it fixes anything. > > Yep, agreed. If this revert kernel doesn't work, we're likely down to a > git bisect, Bruno. Yes, please. I'm currently traveling for KS and Plumbers, so I cannot debug it deeply, unfortunately. If the culprit were really my patch, I still don't figure out why it breaks. In that case, it must be really a subtle timing issue... thanks, Takashi

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Re: [alsa-devel] [PATCH - pcxhr 1/1] ALSA: pcxhr: Add 8 new sound cards
by Takashi Iwai 25 Aug '12

25 Aug '12

At Fri, 24 Aug 2012 16:22:57 +0200, Markus Bollinger wrote: > > add new sound cards VX442HR VX442e PCX442HR PCX442e VX822HR VX822e PCX822HR > and PCX822e > > Signed-off-by: Markus Bollinger <bollinger(a)digigram.com> The changes look OK, but the patch file embedded in the mail looks broken. If it's difficult to embed the patch, could you use an attachment? thanks, Takashi > > diff --git a/pci/pcxhr/pcxhr.c b/pci/pcxhr/pcxhr.c > index e3ac1f7..be4f145 100644 > --- a/pci/pcxhr/pcxhr.c > +++ b/pci/pcxhr/pcxhr.c > @@ -91,6 +91,14 @@ enum { > PCI_ID_PCX924E, > PCI_ID_PCX924HRMIC, > PCI_ID_PCX924E_MIC, > + PCI_ID_VX442HR, > + PCI_ID_PCX442HR, > + PCI_ID_VX442E, > + PCI_ID_PCX442E, > + PCI_ID_VX822HR, > + PCI_ID_PCX822HR, > + PCI_ID_VX822E, > + PCI_ID_PCX822E, > PCI_ID_LAST > }; > > @@ -121,6 +129,14 @@ static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { > { 0x10b5, 0x9056, 0x1369, 0xbb21, 0, 0, PCI_ID_PCX924E, }, > { 0x10b5, 0x9056, 0x1369, 0xbf01, 0, 0, PCI_ID_PCX924HRMIC, }, > { 0x10b5, 0x9056, 0x1369, 0xbf21, 0, 0, PCI_ID_PCX924E_MIC, }, > + { 0x10b5, 0x9656, 0x1369, 0xd001, 0, 0, PCI_ID_VX442HR, }, > + { 0x10b5, 0x9656, 0x1369, 0xd101, 0, 0, PCI_ID_PCX442HR, }, > + { 0x10b5, 0x9056, 0x1369, 0xd021, 0, 0, PCI_ID_VX442E, }, > + { 0x10b5, 0x9056, 0x1369, 0xd121, 0, 0, PCI_ID_PCX442E, }, > + { 0x10b5, 0x9656, 0x1369, 0xd201, 0, 0, PCI_ID_VX822HR, }, > + { 0x10b5, 0x9656, 0x1369, 0xd301, 0, 0, PCI_ID_PCX822HR, }, > + { 0x10b5, 0x9056, 0x1369, 0xd221, 0, 0, PCI_ID_VX822E, }, > + { 0x10b5, 0x9056, 0x1369, 0xd321, 0, 0, PCI_ID_PCX822E, }, > { 0, } > }; > > @@ -160,6 +176,14 @@ static struct board_parameters pcxhr_board_params[] = { > [PCI_ID_PCX924E] = { "PCX924e", 1, 1, 5, 44 }, > [PCI_ID_PCX924HRMIC] = { "PCX924HR-Mic", 1, 1, 5, 44 }, > [PCI_ID_PCX924E_MIC] = { "PCX924e-Mic", 1, 1, 5, 44 }, > +[PCI_ID_VX442HR] = { "VX442HR", 2, 2, 0, 41 }, > +[PCI_ID_PCX442HR] = { "PCX442HR", 2, 2, 0, 41 }, > +[PCI_ID_VX442E] = { "VX442e", 2, 2, 1, 41 }, > +[PCI_ID_PCX442E] = { "PCX442e", 2, 2, 1, 41 }, > +[PCI_ID_VX822HR] = { "VX822HR", 4, 1, 2, 42 }, > +[PCI_ID_PCX822HR] = { "PCX822HR", 4, 1, 2, 42 }, > +[PCI_ID_VX822E] = { "VX822e", 4, 1, 3, 42 }, > +[PCI_ID_PCX822E] = { "PCX822e", 4, 1, 3, 42 }, > }; > > /* boards without hw AES1 and SRC onboard are all using fw_file_set==4 */ > diff --git a/pci/pcxhr/pcxhr_hwdep.c b/pci/pcxhr/pcxhr_hwdep.c > index ec1587c..bf207e3 100644 > --- a/pci/pcxhr/pcxhr_hwdep.c > +++ b/pci/pcxhr/pcxhr_hwdep.c > @@ -66,10 +66,10 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) > err = pcxhr_send_msg(mgr, &rmh); > if (err) > return err; > - /* test 8 or 12 phys out */ > - if ((rmh.stat[0] & MASK_FIRST_FIELD) != mgr->playback_chips * 2) > + /* test 4, 8 or 12 phys out */ > + if ((rmh.stat[0] & MASK_FIRST_FIELD) < mgr->playback_chips * 2) > return -EINVAL; > - /* test 8 or 2 phys in */ > + /* test 4, 8 or 2 phys in */ > if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) < > mgr->capture_chips * 2) > return -EINVAL; > -- > 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm0010: Enable 44.1kHz support
by Mark Brown 25 Aug '12

25 Aug '12

With appropriate clocking configuration the WM0010 driver supports 44.1kHz audio; enable that. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index e1315e0..f820b57 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -722,7 +722,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), }; -#define WM0010_RATES (SNDRV_PCM_RATE_48000) +#define WM0010_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -- 1.7.10.4

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