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[alsa-devel] [PATCH 1/2] ASoC: wm8741: Fix setting interface format for DSP modes
by Axel Lin 18 Oct '11

18 Oct '11

According to the datasheet: Format Control (05h) BITS[3:2] FMT[1:0] Audio data format selection 00 = right justified mode 01 = left justified mode 10 = I2S mode 11 = DSP mode BIT[4] LRP Polarity selec for LRCLK/DSP mode select 0 = normal LRCLK poalrity/DSP mode A 1 = inverted LRCLK poarity/DSP mode B For SND_SOC_DAIFMT_DSP_A, we should set 0x000C instead of 0x0003. For SND_SOC_DAIFMT_DSP_B, we should set 0x001C instead of 0x0013. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8741.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a42b282..85ebe02 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -339,10 +339,10 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0004; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x000C; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x001C; break; default: return -EINVAL; -- 1.7.5.4

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[alsa-devel] [PATCH] MAINTAINERS: Add maintainer for Analog Devices sound CODECs
by Lars-Peter Clausen 18 Oct '11

18 Oct '11

The MAINTAINERS entry for the ADI sound CODEC drivers currently only lists the ADI devices-drivers-devel mailing-list. Add myself as additional contact, since I'm the person at ADI who is currently doing most of the work on these drivers. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- MAINTAINERS | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 0b4ccdd..b2dabb7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -530,6 +530,7 @@ S: Maintained F: drivers/infiniband/hw/amso1100/ ANALOG DEVICES INC ASOC CODEC DRIVERS +M: Lars-Peter Clausen <lars(a)metafoo.de> L: device-drivers-devel(a)blackfin.uclinux.org L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ -- 1.7.2.5

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[alsa-devel] [PATCH v4 2/7] ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
by Ashish Chavan 18 Oct '11

18 Oct '11

This patch add controls for setting cut-off for high pass and voice filters of ADC and DAC. There are also switches to enable/disable these filters. Also removed hard coded, fixed values of these parameters used by previous version of driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 57 +++++++++++++++++++++++++------------------- 1 files changed, 32 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index c7b1635..480fc6d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -167,6 +167,28 @@ static const unsigned int hp_out_tlv[] = { static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -200,6 +222,16 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), }; /* Codec private data */ @@ -275,7 +307,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ @@ -306,68 +337,45 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; - switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -377,7 +385,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); -- 1.7.1

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[alsa-devel] [PATCH v4 1/7] ASoC: da7210: Add support for ADC & DAC equalizers
by Ashish Chavan 18 Oct '11

18 Oct '11

This patch adds support for ADC and DAC five band equalizers available on DA7210 codec. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 37 +++++++++++++++++++++++++++++++++++++ 1 files changed, 37 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index fc99bb7..c7b1635 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -34,10 +34,16 @@ #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_HP_L_VOL 0x21 @@ -158,11 +164,42 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), }; /* Codec private data */ -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Tegra: sparse cleanup
by Olof Johansson 18 Oct '11

18 Oct '11

Fixes the following sparse warnings: sound/soc/tegra/tegra_das.c:215:8: warning: Using plain integer as NULL pointer sound/soc/tegra/tegra_das.c:237:8: warning: Using plain integer as NULL pointer sound/soc/tegra/tegra_pcm.c:370:32: warning: symbol 'tegra_pcm_platform' was not declared. Should it be static? Signed-off-by: Olof Johansson <olof(a)lixom.net> --- sound/soc/tegra/tegra_das.c | 4 ++-- sound/soc/tegra/tegra_pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 9f24ef7..3b55a44 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -212,7 +212,7 @@ err_release: release_mem_region(res->start, resource_size(res)); err_free: kfree(das); - das = 0; + das = NULL; exit: return ret; } @@ -234,7 +234,7 @@ static int __devexit tegra_das_remove(struct platform_device *pdev) release_mem_region(res->start, resource_size(res)); kfree(das); - das = 0; + das = NULL; return 0; } diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c7cfd96..436def1 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -367,7 +367,7 @@ static void tegra_pcm_free(struct snd_pcm *pcm) tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); } -struct snd_soc_platform_driver tegra_pcm_platform = { +static struct snd_soc_platform_driver tegra_pcm_platform = { .ops = &tegra_pcm_ops, .pcm_new = tegra_pcm_new, .pcm_free = tegra_pcm_free, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Convert DA7210 to table based DAPM init
by Mark Brown 18 Oct '11

18 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/da7210.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ff68247..3b5dc0d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -528,9 +528,6 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); - snd_soc_add_controls(codec, da7210_snd_controls, - ARRAY_SIZE(da7210_snd_controls)); - dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); return 0; @@ -542,6 +539,9 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, .volatile_register = da7210_volatile_register, + + .controls = da7210_snd_controls, + .num_controls = ARRAY_SIZE(da7210_snd_controls), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.6.3

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[alsa-devel] [PATCH v3] ASoC: Add support for cs42l73 codec
by Brian Austin 17 Oct '11

17 Oct '11

Signed-off-by:Brian Austin <brian.austin(a)cirrus.com> Signed-off-by:Georgi Vlaev <joe(a)nucleusys.com> This patch adds support for Cirrus Logic CS42L73 low power stereo codec. Applied coding style changes and added correct SoB emails Digital mute to be implemented after more testing --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1076 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 ++++++++++ 4 files changed, 1309 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3449431..9107c48 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -175,6 +176,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 787881b..e89e84b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..609cebf --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1076 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com> + * Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + u32 sysclk; + u8 mclksel; + u32 mclk; + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[] = { +/*0*/ 0x00, 0x42, 0xA7, 0x30, +/*4*/ 0x00, 0x00, 0xF1, 0xDF, +/*8*/ 0x3F, 0x57, 0x53, 0x00, +/*C*/ 0x00, 0x15, 0x00, 0x15, +/*A*/ 0x00, 0x15, 0x00, 0x06, +/*E*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +/*60*/ 0x00, 0x00 +}; + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mixer_text[] = { + "Left", "Right", "Mono Mix"}; + +static const struct soc_enum spk_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum spk_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new dac1_output_mixer[] = { + SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1), + SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1), +}; + +static const struct snd_kcontrol_new dac2_output_mixer[] = { + SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1, 1), + SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3, + 4, 1, 1), + SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1), +}; + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, 6, 1, + 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 1, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum), + SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum), + + SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum), + SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum), + + SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum), + SOC_ENUM("XSP Output Mixer Select", xspout_mixer_enum), + +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), + + SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1), + + SND_SOC_DAPM_MIXER("DAC1 Mixer", SND_SOC_NOPM, 0, 0, + dac1_output_mixer, ARRAY_SIZE(dac1_output_mixer)), + SND_SOC_DAPM_MIXER("DAC2 Mixer", SND_SOC_NOPM, 0, 0, + dac2_output_mixer, ARRAY_SIZE(dac2_output_mixer)), + + SND_SOC_DAPM_PGA("DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_PGA("HP Amp", CS42L73_PWRCTL3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("LO Amp", CS42L73_PWRCTL3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("SPK Amp", CS42L73_PWRCTL3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("EAR Amp", CS42L73_PWRCTL3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOUT"), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + {"HPOUT", NULL, "HP Amp"}, + {"LINEOUT", NULL, "LO Amp"}, + {"SPKOUT", NULL, "SPK Amp"}, + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"HP Amp", NULL, "DAC1"}, + {"LO Amp", NULL, "DAC1"}, + {"SPK Amp", NULL, "DAC2"}, + {"SPKLO Amp", NULL, "DAC2"}, + {"EAR Amp", NULL, "DAC2"}, + + {"DAC1", NULL, "DAC1 Mixer"}, + {"DAC2", NULL, "DAC2 Mixer"}, + + {"DAC1 Mixer", "Headphone Switch", "XSPIN"}, + {"DAC1 Mixer", "LineOut Switch", "XSPIN"}, + {"DAC2 Mixer", "Speakerphone Switch", "XSPIN"}, + {"DAC2 Mixer", "Speakerphone LineOut Switch", "XSPIN"}, + {"DAC2 Mixer", "EarSpeaker Switch", "XSPIN"}, + + {"DAC1 Mixer", "Headphone Switch", "ASPIN"}, + {"DAC1 Mixer", "LineOut Switch", "ASPIN"}, + {"DAC2 Mixer", "Speakerphone Switch", "ASPIN"}, + {"DAC2 Mixer", "Speakerphone LineOut Switch", "ASPIN"}, + {"DAC2 Mixer", "EarSpeaker Switch", "ASPIN"}, + + {"DAC1 Mixer", "Headphone Switch", "VSPIN"}, + {"DAC1 Mixer", "LineOut Switch", "VSPIN"}, + {"DAC2 Mixer", "Speakerphone Switch", "VSPIN"}, + {"DAC2 Mixer", "Speakerphone LineOut Switch", "VSPIN"}, + {"DAC2 Mixer", "EarSpeaker Switch", "VSPIN"}, + + {"PGA Mux Left", "Line A", "LINEINA"}, + {"PGA Mux Right", "Line B", "LINEINB"}, + + {"PGA Mux Left", "Mic 1", "MIC1"}, + {"PGA Mux Right", "Mic 2", "MIC2"}, + + {"PGA Left", NULL, "PGA Mux Left"}, + {"PGA Right", NULL, "PGA Mux Right"}, + + {"ADC Left", "ADC", "PGA Left"}, + {"ADC Right", "ADC", "PGA Right"}, + + {"XSPOUT", NULL, "ADC Left"}, + {"XSPOUT", NULL, "ADC Right"}, + + {"ASPOUT", NULL, "ADC Left"}, + {"ASPOUT", NULL, "ADC Right"}, + + {"VSPOUT", NULL, "ADC Left"}, + {"VSPOUT", NULL, "ADC Right"}, + +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format is supported only in slave mode\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + + } else { + /* CS42L73 Slave */ + dev_dbg(codec->dev, "DAI[%d]: Slave\n", id); + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + cs42l73_update_asrc(codec, id, srate); + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(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: %d\n", ret); + return ret; + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = snd_soc_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = snd_soc_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = snd_soc_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = snd_soc_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = ARRAY_SIZE(cs42l73_reg), + .reg_cache_default = cs42l73_reg, + .reg_word_size = sizeof(u8), + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..188f06d --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Brian Austin <brian.austin(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- 1.7.6.3

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[alsa-devel] [PATCH 0/2] snd-usb-caiaq: Initial Work on NI Maschine Support
by William Light 17 Oct '11

17 Oct '11

These patches extend the existing snd-usb-caiaq driver to support the Native Instruments Maschine controller (http://native-instruments.com/maschine). Future work will be on adding support for the LEDs and the two greyscale screens. Thanks, Will

3 4

[alsa-devel] /dev/dsp open failure
by Appalayagari Sreedhar 17 Oct '11

17 Oct '11

Hi, I have developed a audio driver in ALSA model, the audio ic is interfaced with the TI DM365 processor MCBSP ports. I am working with the 2.6.32 kernel. I am able to play the audio using the aplay command. but when i try to open the device using /dev/dsp i am getting the -ve value. it fails to open the device. when i check the >> ls /dev/dsp it is available in the device directory. I have enabled the Sound OSS related options in the kernel menuconfig. with the same options i am able to open the device for the tlv320aic3x device. Could anyone please suggest where i am doing wrong? and help to solve this problem. One more observation : after doing aplay i am able to open the device using the /dev/dsp. if try to open the device using the /dev/dsp before aplay i am getting the error. could you please tell me where i am doing. -Sreedhar.

2 2

[alsa-devel] snd_pcm_open fails with -16 error
by irfan shaikh 17 Oct '11

17 Oct '11

Hello All, A) In my application i am closing and reopening my device to configure some hw_params for audio frequency change. Close of PCM device is success, but reopen the same device fails I get an below error snd_pcm_open() FAIL with return value -16 {Device or resource busy} How can i avoid above error. {I tried putting usleep between open and close but doesn't work for me} B) I have even tried to configure the HW params in run time to avoid above closing and opening of device. But run time HW params configuartion failed. Is there any way to change HW params at run time ? {My PCM device are in RUNNING state and i want to change HW params at run time} Thanks and regards, Irfan SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 1

Re: [alsa-devel] [pulseaudio-discuss] alsa ucm in pulseaudio
by Mark Brown 17 Oct '11

17 Oct '11

On Fri, Oct 14, 2011 at 12:57:39PM +0200, David Henningsson wrote: > Have you had a look at the patches for UCM already posted several > months ago, and that I spent quite some time reviewing? Or are you > planning to start over from scratch? Yeah, what's the status on those? There didn't seem to be any major issues anyone was raising last time (from memory, I didn't check the archives) and I'm not seeing any clear route to actually getting them merged.

4 5

[alsa-devel] How to save playback out in a file too
by ajay khandelwal 17 Oct '11

17 Oct '11

Hi, I want to same output of playback in a file. We are developing IEC958 device but to debug want to save output of IEC958 plug in a file. Please suggest how to do that? Best Regards, Ajay

2 1

[alsa-devel] Restarting a PCM device
by irfan shaikh 17 Oct '11

17 Oct '11

Hi, Which is the best way to restart an PCM device. If i use a snd_pcm_close() and snd_pcm_open() snd_pcm_open() fails with -16 {device or resource buzzy} error value even snd_pcm_close() is sucessfull. Is it because i am not draining or droping frames before snd_pcm_close ?? Please tell me the best way to restart a PCM device Thanks and regards Irfan SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

1 0

[alsa-devel] [PATCH] ASoC: wm8580: Use snd_soc_update_bits for read-modify-write
by Axel Lin 17 Oct '11

17 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8580.c | 21 +++++++++------------ 1 files changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index b256727..5e348d6 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -430,8 +430,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Always disable the PLL - it is not safe to leave it running * while reprogramming it. */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg | pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, pwr_mask); if (!freq_in || !freq_out) return 0; @@ -449,8 +448,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8580_PLLA4 + offset, reg); /* All done, turn it on */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg & ~pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, 0); return 0; } @@ -757,20 +755,19 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Power up and get individual control of the DACs */ - reg = snd_soc_read(codec, WM8580_PWRDN1); - reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); - snd_soc_write(codec, WM8580_PWRDN1, reg); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN | + WM8580_PWRDN1_ALLDACPD, 0); /* Make VMID high impedance */ - reg = snd_soc_read(codec, WM8580_ADC_CONTROL1); - reg &= ~0x100; - snd_soc_write(codec, WM8580_ADC_CONTROL1, reg); + snd_soc_update_bits(codec, WM8580_ADC_CONTROL1, + 0x100, 0); } break; case SND_SOC_BIAS_OFF: - reg = snd_soc_read(codec, WM8580_PWRDN1); - snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN, WM8580_PWRDN1_PWDN); break; } codec->dapm.bias_level = level; -- 1.7.4.1

1 0

[alsa-devel] Creative X-Fi 51 Pro Support
by Christian Hoff 16 Oct '11

16 Oct '11

Dear all, I just bough and installed a new Creative X-Fi 51 Pro USB sound card. The card works, but it recognized as a full speed 1.1 device instead of high speed (usb 2.0). This causes some limitations - for example I cannot play 5.1ch 24bit/96000Hz audio streams, but only 5.1ch 16/44000Hz. There are various other reports of this behavior in the net(for example [1]), but so far nobody was able to come up with a solution. I would like to fix this in ALSA - probably some kind of vendor-specific quirk is needed which tells the card that it can safely turn on its high-speed capabilities. My assumption is that Creative just disabled the USB high speed features by default because some other OS'es might have problems with audio class 1 devices running at high speed. Now what I would need is some sort of technical input from Creative. A technical documentation of the card or some sample code would be nice. Maybe they are going to send me the specs for their X-Fi 51 Pro if I ask, but the problem is whom at Creative should I ask? Or is there anyone who already has some technical docs for that card available? The reason why I ask is because e.g. the remote control quirks would have required some docs from Creative, so I assume Creative has been sending you technical input in the past. Thank you very much for any help on that matter. Regards, Christian Hoff [1] https://bugs.launchpad.net/ubuntu/+source/linux/+bug/798795/comments/4 -- Empfehlen Sie GMX DSL Ihren Freunden und Bekannten und wir belohnen Sie mit bis zu 50,- Euro! https://freundschaftswerbung.gmx.de

2 1

[alsa-devel] [PATCH v4 4/7] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 15 Oct '11

15 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Reading from register instead of cache - Made debug message more specific about error - Changed "Gain" to "Volume" in names of ALC controls --- sound/soc/codecs/da7210.c | 109 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 109 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cac4f26..0871ebc 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,11 +26,15 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A #define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -59,6 +63,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -81,6 +91,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -150,6 +161,29 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -202,6 +236,68 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EPERM; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != 0xc3) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EPERM; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -261,6 +357,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

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[alsa-devel] [PATCH v3 5/8] ASoC: da7210: Add support for ADC & DAC equalizers
by Ashish Chavan 15 Oct '11

15 Oct '11

This patch adds support for ADC and DAC five band equalizers available on DA7210 codec. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index bc40fd8..8dad77e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -40,10 +40,16 @@ #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_OUT1_L 0x1E @@ -191,6 +197,8 @@ static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -241,6 +249,34 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, inpga_gain_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), }; /* -- 1.7.1

2 2

[alsa-devel] [PATCH v3 1/8] ASoC: da7210: bugfix for head phone volume control
by Ashish Chavan 14 Oct '11

14 Oct '11

This patch takes care of reserved bits of headphone volume register by using correct volume range. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ff68247..0809f32 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -149,12 +149,14 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; static const struct snd_kcontrol_new da7210_snd_controls[] = { -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: wm8990: Use snd_soc_update_bits for read-modify-write
by Axel Lin 14 Oct '11

14 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write This patch also includes a comment fix in wm8990_set_dai_pll(), if freq_in and freq_out are 0, what we do is to clear WM8990_PLL_ENA bit. Thus the comment should be "Turn off PLL". Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8990.c | 67 ++++++++++++++++++--------------------------- 1 files changed, 27 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index b9c5ecc..d29a962 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -981,7 +981,6 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - u16 reg; struct snd_soc_codec *codec = codec_dai->codec; struct _pll_div pll_div; @@ -989,13 +988,12 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, pll_factors(&pll_div, freq_out * 4, freq_in); /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg |= WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, WM8990_PLL_ENA); /* sysclk comes from PLL */ - reg = snd_soc_read(codec, WM8990_CLOCKING_2); - snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_SYSCLK_SRC, WM8990_SYSCLK_SRC); /* set up N , fractional mode and pre-divisor if necessary */ snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | @@ -1003,10 +1001,9 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); snd_soc_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); } else { - /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg &= ~WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, 0); } return 0; } @@ -1084,28 +1081,23 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg; switch (div_id) { case WM8990_MCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_MCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_MCLK_DIV_MASK, div); break; case WM8990_DACCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_DAC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_DAC_CLKDIV_MASK, div); break; case WM8990_ADCCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_ADC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_ADC_CLKDIV_MASK, div); break; case WM8990_BCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_1) & - ~WM8990_BCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_1, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_1, + WM8990_BCLK_DIV_MASK, div); break; default: return -EINVAL; @@ -1164,7 +1156,6 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { int ret; - u16 val; switch (level) { case SND_SOC_BIAS_ON: @@ -1172,9 +1163,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1239,9 +1229,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, } /* VMID=2*250k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x4); break; case SND_SOC_BIAS_OFF: @@ -1255,8 +1244,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, WM8990_BUFIOEN); /* mute DAC */ - val = snd_soc_read(codec, WM8990_DAC_CTRL); - snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + snd_soc_update_bits(codec, WM8990_DAC_CTRL, + WM8990_DAC_MUTE, WM8990_DAC_MUTE); /* Enable any disabled outputs */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); @@ -1344,7 +1333,6 @@ static int wm8990_resume(struct snd_soc_codec *codec) static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - u16 reg; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret < 0) { @@ -1357,15 +1345,14 @@ static int wm8990_probe(struct snd_soc_codec *codec) /* charge output caps */ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4); - snd_soc_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + snd_soc_update_bits(codec, WM8990_AUDIO_INTERFACE_4, + WM8990_ALRCGPIO1, WM8990_ALRCGPIO1); - reg = snd_soc_read(codec, WM8990_GPIO1_GPIO2) & - ~WM8990_GPIO1_SEL_MASK; - snd_soc_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + snd_soc_update_bits(codec, WM8990_GPIO1_GPIO2, + WM8990_GPIO1_SEL_MASK, 1); - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_OPCLK_ENA, WM8990_OPCLK_ENA); snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8400: Fix wrong bit setting for WM8400_POWER_MANAGEMENT_2
by Axel Lin 14 Oct '11

14 Oct '11

If (fakepower & ((1 << WM8400_INMIXR_PWR) | (1 << WM8400_AINRMUX_PWR))) is false, we should clear WM8400_AINR_ENA bits instead of WM8400_AINL_ENA. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8400.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index fbee556..dc13be2 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -383,7 +383,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, (1 << WM8400_AINRMUX_PWR))) { reg |= WM8400_AINR_ENA; } else { - reg &= ~WM8400_AINL_ENA; + reg &= ~WM8400_AINR_ENA; } wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); -- 1.7.4.1

2 2

[alsa-devel] [PATCH 1/2] ASoC: wm8991: Fix wrong bit setting for WM8991_POWER_MANAGEMENT_2
by Axel Lin 14 Oct '11

14 Oct '11

If (fakepower & ((1 << WM8991_INMIXR_PWR_BIT)|(1 << WM8991_AINRMUX_PWR_BIT)))) is false, we should clear WM8991_AINR_ENA bits instead of WM8991_AINL_ENA. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8991.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 08d64a6..708d251 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -393,7 +393,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, (1 << WM8991_AINRMUX_PWR_BIT))) reg |= WM8991_AINR_ENA; else - reg &= ~WM8991_AINL_ENA; + reg &= ~WM8991_AINR_ENA; snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); return 0; -- 1.7.4.1

2 2

[alsa-devel] [PATCH 1/3] ASoC: alc5623: Convert codec->hw_read to snd_soc_read
by Axel Lin 14 Oct '11

14 Oct '11

codec->hw_read is broken now, let's covert to snd_soc_read. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 557b3af..984b14b 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -53,8 +53,10 @@ static void alc5623_fill_cache(struct snd_soc_codec *codec) u16 *cache = codec->reg_cache; /* not really efficient ... */ + codec->cache_bypass = 1; for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = codec->hw_read(codec, i); + cache[i] = snd_soc_read(codec, i); + codec->cache_bypass = 0; } static inline int alc5623_reset(struct snd_soc_codec *codec) -- 1.7.4.1

2 7

[alsa-devel] [PATCH v2] ASoC: tlv320aic23: convert to soc-cache
by Axel Lin 14 Oct '11

14 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: Use snd_soc_cache_write instead of snd_soc_write with cache_only flag to write the register default value to cache for reserved registers. sound/soc/codecs/tlv320aic23.c | 167 ++++++++++++--------------------------- 1 files changed, 52 insertions(+), 115 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3a4bb2..ab27dbc 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -47,63 +47,6 @@ static const u16 tlv320aic23_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ }; -/* - * read tlv320aic23 register cache - */ -static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec - *codec, unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return -1; - return cache[reg]; -} - -/* - * write tlv320aic23 register cache - */ -static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u16 value) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return; - cache[reg] = value; -} - -/* - * write to the tlv320aic23 register space - */ -static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - - u8 data[2]; - - /* TLV320AIC23 has 7 bit address and 9 bits of data - * so we need to switch one data bit into reg and rest - * of data into val - */ - - if (reg > 9 && reg != 15) { - printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); - return -1; - } - - data[0] = (reg << 1) | (value >> 8 & 0x01); - data[1] = value & 0xff; - - tlv320aic23_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - - printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, - value, reg); - - return -EIO; -} - static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -139,8 +82,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, */ val = (val >= 4) ? 4 : (3 - val); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); - tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); + snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } @@ -151,7 +94,7 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val; - val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; @@ -232,7 +175,6 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { enum snd_soc_control_type control_type; - void *control_data; int mclk; int requested_adc; int requested_dac; @@ -344,7 +286,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int src = snd_soc_read(codec, TLV320AIC23_SRATE); int sr = (src >> 2) & 0x0f; int val = (mclk / bosr_usb_divisor_table[src & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; @@ -368,7 +310,7 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + snd_soc_write(codec, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; @@ -407,9 +349,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - iface_reg = - tlv320aic23_read_reg_cache(codec, - TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -423,7 +364,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg |= (0x03 << 2); break; } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -435,7 +376,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; /* set active */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); return 0; } @@ -450,7 +391,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; @@ -463,14 +404,14 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; u16 reg; - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + reg = snd_soc_read(codec, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; - tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + snd_soc_write(codec, TLV320AIC23_DIGT, reg); return 0; } @@ -481,8 +422,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 iface_reg; - iface_reg = - tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -516,7 +456,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -532,26 +472,26 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ TLV320AIC23_DAC_OFF); - tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + snd_soc_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ - TLV320AIC23_CLK_OFF); + snd_soc_write(codec, TLV320AIC23_PWR, + reg | TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->dapm.bias_level = level; @@ -598,13 +538,7 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec, static int tlv320aic23_resume(struct snd_soc_codec *codec) { - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } + snd_soc_cache_sync(codec); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -613,46 +547,52 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); - int reg; + int ret; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec->control_data = aic23->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; - codec->hw_read = NULL; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* Reset codec */ - tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + snd_soc_write(codec, TLV320AIC23_RESET, 0); + + /* Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + snd_soc_cache_write(codec, 0x0A, 0); + snd_soc_cache_write(codec, 0x0B, 0); + snd_soc_cache_write(codec, 0x0C, 0); + snd_soc_cache_write(codec, 0x0D, 0); + snd_soc_cache_write(codec, 0x0E, 0); /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); - tlv320aic23_write(codec, TLV320AIC23_LINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - (TLV320AIC23_LRS_ENABLED)); + snd_soc_update_bits(codec, TLV320AIC23_LINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); - tlv320aic23_write(codec, TLV320AIC23_RINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - TLV320AIC23_LRS_ENABLED); + snd_soc_update_bits(codec, TLV320AIC23_RINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); - tlv320aic23_write(codec, TLV320AIC23_ANLG, - (reg) & (~TLV320AIC23_BYPASS_ON) & - (~TLV320AIC23_MICM_MUTED)); + snd_soc_update_bits(codec, TLV320AIC23_ANLG, + TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, + 0); /* Default output volume */ - tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); @@ -674,8 +614,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, - .read = tlv320aic23_read_reg_cache, - .write = tlv320aic23_write, .set_bias_level = tlv320aic23_set_bias_level, .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), @@ -702,7 +640,6 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, aic23); - aic23->control_data = i2c; aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

2 1

[alsa-devel] [PATCH v3 3/8] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 14 Oct '11

14 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed explicit setting of default gains - Renamed "DAC Playback" to "Digital Playback" --- sound/soc/codecs/da7210.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 43 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index b6e5085..a7ea07d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -42,6 +42,8 @@ #define DA7210_DAC_SEL 0x17 #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F #define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 @@ -141,6 +143,17 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -161,11 +174,35 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); + +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), }; /* @@ -659,6 +696,12 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: max98095: Convert codec->hw_write to snd_soc_write
by Axel Lin 14 Oct '11

14 Oct '11

codec->hw_write is broken now, convert codec->hw_write to snd_soc_write. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/max98095.c | 26 ++++---------------------- 1 files changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6982f74..85f9379 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -610,24 +610,6 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) } /* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - data[0] = reg; - data[1] = value; - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -/* * Load equalizer DSP coefficient configurations registers */ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, @@ -647,8 +629,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -672,8 +654,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } -- 1.7.4.1

3 2

[alsa-devel] [PATCH 0/2] Conexant HD-audio fixes
by Takashi Iwai 14 Oct '11

14 Oct '11

Hi, while debugging a problem reported by a Thinkpad user, we found that there are fundamental problems in the current Conexant auto-parser code. 1. EAPDs on older chips don't correspond to the pin as 1:1. A single EAPD can influence on the whole outputs, so we can't turn it off simply even when the assigned pin isn't used. 2. There is only a single input-amp in audio-in widgets. This is contradicting against what I was informed before. There is only one capture level in each audio-in widget no matter how many sources are connected to it. For example, if an ADC has the connection to an internal mic and an external mic, the volume in ADC is only one, no individual amp per connection, unlike other widgets. For the first issue, a fix is just keep EAPD turned on for such chips. This simplifies the previous David's fix. About the second part, I'm not 100% sure whether it's correct to all Conexant chips. But this sounds more natural, and I always wondered about the old implementation. Can anyone confirm that the patches work properly? The initial test was done with a machine with cx20549, so tests with other codecs like cx20551, cx20561, and cx2058x would be appreciated. After applying the patch, load snd-hda-intel with model=auto option. You'll have a single "Capture" volume (or two capture volumes like "Mic" and "Internal Mic", which correspond to different ADCs). Check the different recording source and whether the capture volume(s) really changes the recording level properly. thanks, Takashi

1 2

[alsa-devel] [PATCH v2 2/9] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 14 Oct '11

14 Oct '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 54 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 47 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 1ef058b..87b9ae5 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,10 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -147,6 +151,15 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_EN (1 << 7) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) + #define DA7210_VERSION "0.0.1" /* @@ -165,6 +178,9 @@ */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -180,6 +196,13 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -193,6 +216,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + + /* MIC related controls */ + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + + /* AUX related controls */ + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), }; /* Codec private data */ @@ -246,13 +284,9 @@ static int da7210_startup(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + /* Enable Mic,AUX1 and AUX2 */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD); } return 0; @@ -521,6 +555,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: WM8903: Implement DMIC support
by Stephen Warren 14 Oct '11

14 Oct '11

In addition to the currently supported analog capture path, the WM8903 also supports digital mics. The analog and digital capture paths are exclusive; a mux is present to select the capture source. Logically, the mux exists to select the decimator's input, from either the ADC or DMIC block outputs. However, the ADC power domain also includes the DMIC interface. Consequently, this change represents the mux as existing immediately before the ADC, and selecting between the Input PGA and DMIC block outputs. An alternative might be to represent the mux in its correct location, and associate the ADC power enable controls with both the real ADC, and a fake ADC for the DMIC? Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d53f206..f3cab84 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -634,6 +634,13 @@ static const struct soc_enum lsidetone_enum = static const struct soc_enum rsidetone_enum = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static const char *adcinput_text[] = { + "ADC", "DMIC" +}; + +static const struct soc_enum adcinput_enum = + SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text); + static const char *aif_text[] = { "Left", "Right" }; @@ -767,6 +774,9 @@ static const struct snd_kcontrol_new lsidetone_mux = static const struct snd_kcontrol_new rsidetone_mux = SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); +static const struct snd_kcontrol_new adcinput_mux = + SOC_DAPM_ENUM("ADC Input", adcinput_enum); + static const struct snd_kcontrol_new lcapture_mux = SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum); @@ -817,6 +827,7 @@ SND_SOC_DAPM_INPUT("IN2L"), SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), @@ -842,6 +853,9 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux), SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), +SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), +SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), + SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0), @@ -979,6 +993,11 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Left Input PGA", NULL, "Left Input Mode Mux" }, { "Right Input PGA", NULL, "Right Input Mode Mux" }, + { "Left ADC Input", "ADC", "Left Input PGA" }, + { "Left ADC Input", "DMIC", "DMICDAT" }, + { "Right ADC Input", "ADC", "Right Input PGA" }, + { "Right ADC Input", "DMIC", "DMICDAT" }, + { "Left Capture Mux", "Left", "ADCL" }, { "Left Capture Mux", "Right", "ADCR" }, @@ -988,9 +1007,9 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "AIFTXL", NULL, "Left Capture Mux" }, { "AIFTXR", NULL, "Right Capture Mux" }, - { "ADCL", NULL, "Left Input PGA" }, + { "ADCL", NULL, "Left ADC Input" }, { "ADCL", NULL, "CLK_DSP" }, - { "ADCR", NULL, "Right Input PGA" }, + { "ADCR", NULL, "Right ADC Input" }, { "ADCR", NULL, "CLK_DSP" }, { "Left Playback Mux", "Left", "AIFRXL" }, -- 1.7.1

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Re: [alsa-devel] [pulseaudio-discuss] alsa ucm in pulseaudio
by Mark Brown 14 Oct '11

14 Oct '11

On Fri, Oct 14, 2011 at 12:19:53PM +0200, David Henningsson wrote: > On 10/14/2011 11:42 AM, Feng Wei wrote: > >Actually we must map the concepts because ucm is designed for abstract > >the complicated kcontrol. In my mind, if we use ucm in pa, the > >original profile configuration files and mixer configuration files > >will be replaced by ucm configuration files. > Removing the original profile configuration files might be possible > in an embedded scenario, but for all hundreds of different > snd-hda-intel/AC'97/etc configurations out there, writing UCM > configurations to fit all of them is just not scalable. (If so, one > would have to write an auto-UCM tool that autogenerates UCM profiles > based on present hardware.) Yes, writing UCM files for PCs which have reasonably regular and discoverable audio is silly - if UCM is used at all on most desktops it should be with automatically generated configuration.

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Re: [alsa-devel] [pulseaudio-discuss] alsa ucm in pulseaudio
by Mark Brown 14 Oct '11

14 Oct '11

On Fri, Oct 14, 2011 at 10:57:08AM +0200, David Henningsson wrote: > There is also the "Use Case" concept in UCM, Is there always exactly > one verb for every use case? If not, one might wonder if "Use Case" > or "Verb" is what should correspond to the "Profile" concept. Yes, pretty much. > As for ports, this again depends on what is mutually exclusive and > what could be used in parallel, I vaguely remember "Modifiers" as > something that could be used in parallel with existing streams, if > so, they need to be separate sinks/sources. Modifiers and Devices can both be used concurrently. > Hmm, for every sink/source, there can only be one active port at a > time. Also, ports on the same sink/source can be changed on the fly > without restarting the stream. Does this fit with the description of > a UCM "Device"? Pretty much.

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[alsa-devel] [PATCH v3 8/8] ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
by Ashish Chavan 14 Oct '11

14 Oct '11

This patch add controls for setting cut-off for high pass and voice filters of ADC and DAC. There are also switches to enable/disable these filters. Also removed hard coded, fixed values of these parameters used by previous version of driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 57 +++++++++++++++++++++++++------------------- 1 files changed, 32 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index f2e154a..ecb4cb7 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -260,6 +260,28 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -407,6 +429,16 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), }; /* @@ -638,7 +670,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ @@ -669,68 +700,45 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; - switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -740,7 +748,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); -- 1.7.1

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[alsa-devel] [PATCH v3 7/8] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 14 Oct '11

14 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Reading from register instead of cache - Made debug message more specific about error - Changed "Gain" to "Volume" in names of ALC controls --- sound/soc/codecs/da7210.c | 102 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 102 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ef67a4d..f2e154a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_STARTUP2 0x04 @@ -67,6 +68,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -89,6 +96,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -167,9 +175,11 @@ #define DA7210_OUT2_EN (1 << 7) /* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) #define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) #define DA7210_AUX1_R_EN (1 << 7) /* AUX2 bit fields */ @@ -178,6 +188,23 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -233,6 +260,68 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EPERM; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != 0xc3) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EPERM; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -305,6 +394,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* -- 1.7.1

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[alsa-devel] [PATCH v3 6/8] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 14 Oct '11

14 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8dad77e..ef67a4d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -47,6 +47,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -58,6 +59,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -173,6 +175,9 @@ /* AUX2 bit fields */ #define DA7210_AUX2_EN (1 << 3) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -221,6 +226,13 @@ static const unsigned int aux1_vol_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) }; +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -277,6 +289,22 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Capture Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Capture Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* @@ -674,6 +702,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -681,6 +721,7 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) static struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -782,6 +823,9 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] busy loop in snd_pcm_plugin_status()
by Thierry Reding 14 Oct '11

14 Oct '11

Hi, I'm seeing a strange issue when using linphone (3.4.3) with alsa-lib (1.0.24.1) on an Atom-based platform. I was able to trace the problem down to the snd_pcm_plugin_status() function. The problem is that the call to the snd_atomic_read_ok() function fails because snd_pcm_plugin_avail_update() has already called snd_atomic_write_{begin,end}() in the meantime. The result is that I'm seeing calls to snd_pcm_plugin_status() where the code jumps back to the _again between 200 and 1800 times before succeeding. This in turn leads to delays in the linphone of 20 to 180 milliseconds, which results in rather absurdly high CPU usage (usually 100%) and correspondingly bad audio playback and side-effects (choppiness, linphone no longer terminates calls properly, ...). I'm using the attached patch as a workaround, but I'm pretty sure it's not a proper fix. The strange thing is that I see that behaviour only on the Atom platform (Intel HDA, Sigmatel 92HD83 codec). Another ARM-based platform does not exhibit the same issue. Can anybody shed some light on the situation? Cheers, Thierry

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[alsa-devel] [PATCH v3 4/8] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 14 Oct '11

14 Oct '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a7ea07d..bc40fd8 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -32,6 +32,10 @@ #define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -154,6 +158,15 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_EN (1 << 7) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) + #define DA7210_VERSION "0.0.1" /* @@ -175,6 +188,9 @@ static const unsigned int hp_out_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -190,6 +206,13 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -203,6 +226,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + + /* MIC related controls */ + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + + /* AUX related controls */ + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), }; /* @@ -702,6 +740,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] [RFC 0/2] ASoC: core: Ignoring pmdown_time for playback stream power down
by Peter Ujfalusi 14 Oct '11

14 Oct '11

Hello Mark, Add support in core to ignore the pmdown_time when stopping the playback stream. This will inline the DAPM stream stop event, so it is going to be executed without delay at stream close time (for playback stream). Backround: with the McPDM protocol we are sendning not only the pure audio stream, but OMAP McPDM also transmits additional information (for example offset cancellation). If McPDM is stopped prior to the DAC this information will be not sent to the codec, which can result noise rendered by the twl6040 codec. Is this something that you were suggesting? Regards, Peter --- Peter Ujfalusi (2): ASoC: soc-pcm: Add flag to ignore pmdown_time at pcm_close ASoC: sdp4430: Request core to inline the DAPM sequence include/sound/soc.h | 3 +++ sound/soc/omap/sdp4430.c | 1 + sound/soc/soc-pcm.c | 15 +++++++++++---- 3 files changed, 15 insertions(+), 4 deletions(-) -- 1.7.7

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[alsa-devel] ASoC: mixing PCI card and SoC codec
by Matteo Fortini 14 Oct '11

14 Oct '11

I have a Geode board with integrated PCI CS5535 sound card, and I'll be using an external ADAU1761 codec as an aux_dev (mixer and probably DSP). Until now I defined the ADAU as a dummy sound card, just to be able to use its mixer. I guess there's a cleaner solution to that, which I suspect involves bridging SoC and PCI codecs (I think avoiding to call snd_device_new in the PCI codec and moving it to a SOC card would be a good start...). Should I try that, or stick to the dummy soc implementation? Thank you.

1 0

[alsa-devel] [PATCH 1/3] ASoC: alc5623: Convert codec->hw_read to regmap_read
by Axel Lin 14 Oct '11

14 Oct '11

codec->hw_read is broken now, let's covert to regmap_read. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 557b3af..8c156cb 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -51,10 +51,13 @@ static void alc5623_fill_cache(struct snd_soc_codec *codec) { int i, step = codec->driver->reg_cache_step; u16 *cache = codec->reg_cache; + unsigned int val; /* not really efficient ... */ - for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = codec->hw_read(codec, i); + for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) { + regmap_read(codec->control_data, i, &val); + cache[i] = val; + } } static inline int alc5623_reset(struct snd_soc_codec *codec) -- 1.7.4.1

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Re: [alsa-devel] linux-next: Tree for Oct 12 (sound/soc/codecs/wm1250-ev1)
by Randy Dunlap 14 Oct '11

14 Oct '11

On 10/11/2011 11:54 PM, Stephen Rothwell wrote: > Hi all, > > The linux-next tree is now available from > git://github.com/sfrothwell/linux-next.git as a temporary measure while > the kernel.org servers are unavailable. > > It may also turn up on git.kernel.org (depending on the mirroring). The > patch is still absent, however. When no GPIO kconfig options are enabled: (not GPIOLIB, not GENERIC_GPIO): sound/soc/codecs/wm1250-ev1.c:32:14: error: array type has incomplete element type sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:126:18: error: negative width in bit-field '<anonymous>' sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:134:42: error: negative width in bit-field '<anonymous>' sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to 'int' in type name sound/soc/codecs/wm1250-ev1.c:155:34: error: negative width in bit-field '<anonymous>' This was on an i386 build, but that shouldn't matter. -- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Use snd_soc_update_bits for read-modify-write
by Axel Lin 13 Oct '11

13 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch also fixes the mix usage of AIC32X4_MDACEN/AIC32X4_NDACEN/AIC32X4_NADCEN/AIC32X4_MADCEN in current code. (They are all defined as (0x01 << 7), so it is not a bug, but not good for readability.) Axel sound/soc/codecs/tlv320aic32x4.c | 61 +++++++++++++++----------------------- 1 files changed, 24 insertions(+), 37 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index a68982e..b21c610 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -528,40 +528,33 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - u8 value; switch (level) { case SND_SOC_BIAS_ON: if (aic32x4->master) { /* Switch on PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value | AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); /* Switch on NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value | AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); /* Switch on MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); /* Switch on NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); /* Switch on MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value | AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); /* Switch on BCLK_N Divider */ - value = snd_soc_read(codec, AIC32X4_BCLKN); - snd_soc_write(codec, AIC32X4_BCLKN, - value | AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); } break; case SND_SOC_BIAS_PREPARE: @@ -569,34 +562,28 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (aic32x4->master) { /* Switch off PLL */ - value = snd_soc_read(codec, AIC32X4_PLLPR); - snd_soc_write(codec, AIC32X4_PLLPR, - (value & ~AIC32X4_PLLEN)); + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); /* Switch off NDAC Divider */ - value = snd_soc_read(codec, AIC32X4_NDAC); - snd_soc_write(codec, AIC32X4_NDAC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); /* Switch off MDAC Divider */ - value = snd_soc_read(codec, AIC32X4_MDAC); - snd_soc_write(codec, AIC32X4_MDAC, - value & ~AIC32X4_MDACEN); + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); /* Switch off NADC Divider */ - value = snd_soc_read(codec, AIC32X4_NADC); - snd_soc_write(codec, AIC32X4_NADC, - value & ~AIC32X4_NDACEN); + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); /* Switch off MADC Divider */ - value = snd_soc_read(codec, AIC32X4_MADC); - snd_soc_write(codec, AIC32X4_MADC, - value & ~AIC32X4_MDACEN); - value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); /* Switch off BCLK_N Divider */ - snd_soc_write(codec, AIC32X4_BCLKN, - value & ~AIC32X4_BCLKEN); + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); } break; case SND_SOC_BIAS_OFF: -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2 5/9] ASoC: da7210: Add support for other DAI word lengths, format and mode
by Ashish Chavan 13 Oct '11

13 Oct '11

This patchs adds support for following, (1) DAI 20 and 32 bit word sizes (2) DAI left and right justified formats (3) DAI slave mode Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 23 ++++++++++++++++++++++- 1 files changed, 22 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cb00935..8cbeb6c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -120,12 +120,17 @@ /* DAI_CFG1 bit fields */ #define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S20_3LE (1 << 0) #define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_WORD_S32_LE (3 << 0) #define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_SLAVE (0 << 7) #define DA7210_DAI_MODE_MASTER (1 << 7) /* DAI_CFG3 bit fields */ #define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_FORMAT_LEFT_J (1 << 0) +#define DA7210_DAI_FORMAT_RIGHT_J (2 << 0) #define DA7210_DAI_OE (1 << 3) #define DA7210_DAI_EN (1 << 7) @@ -383,9 +388,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; + break; case SNDRV_PCM_FORMAT_S24_LE: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_cfg1 |= DA7210_DAI_WORD_S32_LE; + break; default: return -EINVAL; } @@ -490,6 +501,9 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_CBM_CFM: dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_cfg1 |= DA7210_DAI_MODE_SLAVE; + break; default: return -EINVAL; } @@ -502,6 +516,12 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_I2S: dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; break; + case SND_SOC_DAIFMT_LEFT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J; + break; default: return -EINVAL; } @@ -518,7 +538,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } -#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { -- 1.7.1

2 1

[alsa-devel] [PATCH v2 4/9] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX, ADC and DAC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7e36ed6..cb00935 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -45,6 +45,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -56,6 +57,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -166,6 +168,9 @@ /* AUX2 bit fields */ #define DA7210_AUX2_EN (1 << 3) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -211,6 +216,13 @@ static const unsigned int aux1_vol_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) }; +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -267,6 +279,23 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Mute Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Mute Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Mute Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Mute Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("DAC Mute Switch", DA7210_DAC_HPF, 2, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("DAC Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -597,6 +626,9 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: twl6040: Remove PLL restrictions
by Peter Ujfalusi 13 Oct '11

13 Oct '11

Hello, twl6040 has no restrictions on the selected PLL, and sysclk. All input, and output paths can be used with any PLL configuration. Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: Remove Capture restriction for 17.64MHz sysclk ASoC: twl6040: Remove PLL usage restrictions ASoC: twl6040: Change event ordering for Earphone driver ASoC: twl6040: Rename the Earphone Driver event handler sound/soc/codecs/twl6040.c | 53 ++++++++++---------------------------------- 1 files changed, 12 insertions(+), 41 deletions(-) -- 1.7.7

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[alsa-devel] [PATCH v2 8/9] ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch add controls for setting cut-off for high pass and voice filters of ADC and DAC. There are also switches to enable/disable these filters. Also removed hard coded, fixed values of these parameters used by previous version of driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 57 +++++++++++++++++++++++++------------------- 1 files changed, 32 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8df7d84..fcbe557 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -258,6 +258,28 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -406,6 +428,16 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), }; /* Codec private data */ @@ -477,7 +509,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ @@ -508,68 +539,45 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; - switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -579,7 +587,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); -- 1.7.1

1 0

[alsa-devel] [PATCH v2 7/9] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Reading from register instead of cache - Made debug message more specific about error - Changed "Gain" to "Volume" in names of ALC controls --- sound/soc/codecs/da7210.c | 113 +++++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 109 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8cbeb6c..8df7d84 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 @@ -65,6 +66,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -87,6 +94,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -165,9 +173,11 @@ #define DA7210_OUT2_EN (1 << 7) /* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) #define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) #define DA7210_AUX1_R_EN (1 << 7) /* AUX2 bit fields */ @@ -176,6 +186,23 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -187,12 +214,15 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); @@ -228,6 +258,68 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EPERM; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != 0xc3) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EPERM; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -301,6 +393,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

1 0

[alsa-devel] [PATCH v2 6/9] ASoC: da7210: bugfix for head phone volume control
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch takes care of reserved bits of headphone volume register by using correct volume range. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8cbeb6c..21ef6e0 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -187,12 +187,15 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); -- 1.7.1

1 0

[alsa-devel] [PATCH v2 3/9] ASoC: da7210: Add support for ADC & DAC equalizers
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds support for ADC and DAC five band equalizers available on DA7210 codec. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 87b9ae5..7e36ed6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -38,10 +38,16 @@ #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_OUT1_L 0x1E @@ -181,6 +187,8 @@ static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -231,6 +239,34 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, inpga_gain_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), }; /* Codec private data */ -- 1.7.1

1 0

[alsa-devel] [PATCH v2 1/9] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 13 Oct '11

13 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed explicit setting of default gains - Renamed "DAC Playback" to "Digital Playback" --- sound/soc/codecs/da7210.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 43 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a9d9d39..1ef058b 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -40,6 +40,9 @@ #define DA7210_DAC_SEL 0x17 #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -133,6 +136,17 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -150,12 +164,35 @@ * -> min = -79.5 dB */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); + +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), }; /* Codec private data */ @@ -478,6 +515,12 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

1 0

[alsa-devel] [PATCH] ASoC: ak4535: fixup cache register table
by Axel Lin 13 Oct '11

13 Oct '11

ak4535_reg should be 8bit, but cache table is defined as 16bit. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4535.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 68df32d..95d782d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -39,11 +39,11 @@ struct ak4535_priv { /* * ak4535 register cache */ -static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { - 0x0000, 0x0080, 0x0000, 0x0003, - 0x0002, 0x0000, 0x0011, 0x0001, - 0x0000, 0x0040, 0x0036, 0x0010, - 0x0000, 0x0000, 0x0057, 0x0000, +static const u8 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x00, 0x80, 0x00, 0x03, + 0x02, 0x00, 0x11, 0x01, + 0x00, 0x40, 0x36, 0x10, + 0x00, 0x00, 0x57, 0x00, }; static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: ak4642: convert to soc-cache
by Axel Lin 13 Oct '11

13 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi Kuninori, This patch converts ak4642 to use soc-cache APIs. I don't have this hardware, I'd appreciate if you can test this patch. Thanks, Axel sound/soc/codecs/ak4642.c | 82 ++++++-------------------------------------- 1 files changed, 12 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 68fc16c..80a9fe8 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -157,7 +157,6 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { struct ak4642_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -176,64 +175,6 @@ static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { 0x0000, }; -/* - * read ak4642 register cache - */ -static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4642 register cache - */ -static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return; - - cache[reg] = value; -} - -/* - * write to the AK4642 register space - */ -static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4642 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (codec->hw_write(codec->control_data, data, 2) == 2) { - ak4642_write_reg_cache(codec, reg, value); - return 0; - } else - return -EIO; -} - -static int ak4642_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4642_CACHEREGNUM; i++) - r |= ak4642_write(codec, i, cache[i]); - - return r; -}; - static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -253,8 +194,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, */ snd_soc_update_bits(codec, MD_CTL4, DACH, DACH); snd_soc_update_bits(codec, MD_CTL3, BST1, BST1); - ak4642_write(codec, L_IVC, 0x91); /* volume */ - ak4642_write(codec, R_IVC, 0x91); /* volume */ + snd_soc_write(codec, L_IVC, 0x91); /* volume */ + snd_soc_write(codec, R_IVC, 0x91); /* volume */ snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC, PMVCM | PMMIN | PMDAC); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); @@ -273,9 +214,9 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - ak4642_write(codec, SG_SL1, PMMP | MGAIN0); - ak4642_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); - ak4642_write(codec, ALC_CTL1, ALC | LMTH0); + snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL, PMVCM | PMADL); snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); @@ -463,7 +404,7 @@ static struct snd_soc_dai_driver ak4642_dai = { static int ak4642_resume(struct snd_soc_codec *codec) { - ak4642_sync(codec); + snd_soc_cache_sync(codec); return 0; } @@ -471,11 +412,15 @@ static int ak4642_resume(struct snd_soc_codec *codec) static int ak4642_probe(struct snd_soc_codec *codec) { struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); - codec->hw_write = (hw_write_t)i2c_master_send; - codec->control_data = ak4642->control_data; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } snd_soc_add_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); @@ -486,8 +431,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, .resume = ak4642_resume, - .read = ak4642_read_reg_cache, - .write = ak4642_write, .reg_cache_size = ARRAY_SIZE(ak4642_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, @@ -505,7 +448,6 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4642); - ak4642->control_data = i2c; ak4642->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

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[alsa-devel] [RFC][PATCH] ASoC: Suppress restore of default register values for lzo cache sync
by Axel Lin 13 Oct '11

13 Oct '11

Currently we do suppress restore of default register values for rbtree and flat cache sync. A couple users use this trick to avoid writing reset register while resume. Thus make lzo cache sync suppress restore of default register values, so we have consistent behavior for all types of cache sync. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/soc-cache.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9077aa4..eb3c016 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -555,16 +555,22 @@ static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) { struct snd_soc_lzo_ctx **lzo_blocks; - unsigned int val; + unsigned int val, def; + unsigned int word_size; int i; int ret; lzo_blocks = codec->reg_cache; + word_size = codec->driver->reg_word_size; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { WARN_ON(!snd_soc_codec_writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; + def = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size); + if (val == def) + continue; + codec->cache_bypass = 1; ret = snd_soc_write(codec, i, val); codec->cache_bypass = 0; -- 1.7.4.1

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 13 Oct '11

13 Oct '11

The following changes since commit 03b5362d2fef02cabbbb440d861d27e40ed3a929: ASoC: Convert Jive to table based init (2011-10-10 11:10:44 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Ashish Chavan (1): ASoC: da7210: Add support for other DAI word lengths, format and mode Axel Lin (6): ASoC: Writing register default value for the reset register ASoC: ak4642: convert to soc-cache ASoC: Delete ads117x.h ASoC: Remove impossible case from wm8994_hw_params ASoC: da7210: convert to soc-cache ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X Mark Brown (4): ASoC: Convert Goni to data based DAPM init ASoC: Add missing default for WM5100 Clocking 1 ASoC: Implement WM5100 accessory detection support ASoC: Instantiate card widgets immediately Peter Ujfalusi (18): ASoC: OMAP machines: Remove soc_dapm_sync() call from init ASoC: zoom2: Let core to deal with the DAPM widgets ASoC: zoom2: No need to call dapm_pin_enable at init time ASoC: sdp4430: Let core to deal with the DAPM widgets ASoC: sdp4430: No need to call dapm_pin_enable at init time ASoC: am3517evm: Let core to deal with the DAPM widgets ASoC: n810: Let the core to register DAPM widgets/routes and controls ASoC: osk5912: Let core to deal with the DAPM widgets ASoC: sdp3430: Let core to deal with the DAPM widgets ASoC: twl6040: Convert to table based init ASoC: twl4030: Convert to table based init ASoC: tlv320dac33: Convert to table based init Input: twl6040: Simplify vibra regsiter definitions MFD: twl6040: Cache the vibra control registers MFD: twl6040: function to query the vibra status for clients Input: twl6040-vibra: Check the selected path for vibra ASoC: twl6040: Support for vibra output paths ASoC: twl6040: Workaround for headset DC offset caused pop noise Wolfram Sang (1): ASoC: Fix typo in Kconfig symbol for tlv320aic32x4 drivers/input/misc/twl6040-vibra.c | 19 +++-- drivers/mfd/twl6040-core.c | 31 ++++++- include/linux/mfd/twl6040.h | 25 +++--- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/ads117x.h | 13 --- sound/soc/codecs/ak4642.c | 82 +++---------------- sound/soc/codecs/da7210.c | 146 +++++++++++++++----------------- sound/soc/codecs/tlv320dac33.c | 23 ++---- sound/soc/codecs/twl4030.c | 21 ++--- sound/soc/codecs/twl6040.c | 135 ++++++++++++++++++++++++------ sound/soc/codecs/wm5100-tables.c | 1 + sound/soc/codecs/wm5100.c | 162 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm5100.h | 2 + sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8994.c | 10 -- sound/soc/imx/Kconfig | 2 +- sound/soc/omap/am3517evm.c | 28 +----- sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/n810.c | 23 ++---- sound/soc/omap/omap3pandora.c | 8 +-- sound/soc/omap/osk5912.c | 27 +----- sound/soc/omap/rx51.c | 2 - sound/soc/omap/sdp3430.c | 18 +--- sound/soc/omap/sdp4430.c | 27 +----- sound/soc/omap/zoom2.c | 26 ++----- sound/soc/pxa/Kconfig | 2 + sound/soc/samsung/goni_wm8994.c | 13 +-- sound/soc/soc-core.c | 2 + 29 files changed, 467 insertions(+), 390 deletions(-) delete mode 100644 sound/soc/codecs/ads117x.h

2 1

[alsa-devel] [PATCH] ASoC: sta32x: Set reg_cache_default to sta32x_regs
by Axel Lin 13 Oct '11

13 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sta32x.c | 5 +---- 1 files changed, 1 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 5c7def3..754b3ff 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -756,10 +756,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* read reg reset values into cache */ - for (i = 0; i < STA32X_REGISTER_COUNT; i++) - snd_soc_cache_write(codec, i, sta32x_regs[i]); - /* preserve reset values of reserved register bits */ snd_soc_cache_write(codec, STA32X_CONFC, codec->hw_read(codec, STA32X_CONFC)); @@ -837,6 +833,7 @@ static const struct snd_soc_codec_driver sta32x_codec = { .resume = sta32x_resume, .reg_cache_size = STA32X_REGISTER_COUNT, .reg_word_size = sizeof(u8), + .reg_cache_default = sta32x_regs, .volatile_register = sta32x_reg_is_volatile, .set_bias_level = sta32x_set_bias_level, .controls = sta32x_snd_controls, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: pxa: Remove redundant snd_soc_dapm_sync() calls from machine drivers
by Axel Lin 13 Oct '11

13 Oct '11

The core will sync DAPM as part of the card initialization, there is no need for machine drivers to do so during their setup. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/poodle.c | 1 - sound/soc/pxa/spitz.c | 1 - 2 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index da3ae43..4c29bc1 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -265,7 +265,6 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ce920e3..c2d6ff9 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -301,7 +301,6 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } -- 1.7.4.1

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[alsa-devel] [PATCH 7/9] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 104 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 104 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index eb12c82..9cab62a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 @@ -65,6 +66,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -87,6 +94,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -165,9 +173,11 @@ #define DA7210_OUT2_EN (1 << 7) /* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) #define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) #define DA7210_AUX1_R_EN (1 << 7) /* AUX2 bit fields */ @@ -176,6 +186,19 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + /* Default gain/vol values */ #define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ #define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ @@ -185,6 +208,10 @@ #define DA7210_DFLT_AUX2_VOL 0x02 /* 6dB */ #define DA7210_DFLT_INPGA_VOL 0x07 /* 6dB */ +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -247,6 +274,70 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 *reg_cache = codec->reg_cache; + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (reg_cache[DA7210_CONTROL] & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, "Failed to enable ALC\n"); + return -EPERM; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 *reg_cache = codec->reg_cache; + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (reg_cache[DA7210_ADC] & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((reg_cache[DA7210_ZERO_CROSS] & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != 0xc3) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = reg_cache[DA7210_IN_GAIN]; + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((reg_cache[DA7210_AUX1_L] & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((reg_cache[DA7210_AUX1_R] & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + dev_dbg(codec->dev, "Failed to enable Noise suppression\n"); + return -EPERM; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -322,6 +413,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Gain", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Gain", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Gain", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

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[alsa-devel] [PATCH 6/9] ASoC: da7210: bugfix for head phone volume control
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch takes care of reserved bits of headphone volume register by using correct volume range. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 60f403b..eb12c82 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -196,12 +196,15 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); -- 1.7.1

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[alsa-devel] [PATCH 3/9] ASoC: da7210: Add support for ADC & DAC equalizers
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds support for ADC and DAC five band equalizers available on DA7210 codec. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 6039ada..b48a8af 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -38,10 +38,16 @@ #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_OUT1_L 0x1E @@ -190,6 +196,8 @@ static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -248,6 +256,34 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, inpga_gain_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), }; /* Codec private data */ -- 1.7.1

2 2

[alsa-devel] [PATCH 2/9] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 13 Oct '11

13 Oct '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 79 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 70 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index eef861d..6039ada 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,10 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -147,10 +151,23 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_EN (1 << 7) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) + /* Default gain/vol values */ #define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ #define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ #define DA7210_DFLT_OUT2_VOL 0x06 /* 0dB */ +#define DA7210_DFLT_MIC_VOL 0x02 /* 6dB */ +#define DA7210_DFLT_AUX1_VOL 0x35 /* 6dB */ +#define DA7210_DFLT_AUX2_VOL 0x02 /* 6dB */ +#define DA7210_DFLT_INPGA_VOL 0x07 /* 6dB */ #define DA7210_VERSION "0.0.1" @@ -170,6 +187,9 @@ */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -185,6 +205,20 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + +static const char *da7210_mic_bias_voltage_txt[] = { + "1.5", "1.6", "2.2", "2.3" +}; + +static const struct soc_enum da7210_mic_bias = + SOC_ENUM_SINGLE(DA7210_MIC_L, 4, 4, da7210_mic_bias_voltage_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -198,6 +232,22 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + + /* MIC related controls */ + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_ENUM("Mic Bias Voltage", da7210_mic_bias), + + /* AUX related controls */ + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), }; /* Codec private data */ @@ -251,13 +301,9 @@ static int da7210_startup(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + /* Enable Mic,AUX1 and AUX2 */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD); } return 0; @@ -492,8 +538,10 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* Enable Left & Right MIC PGA and Mic Bias */ - snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); - snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN | + DA7210_DFLT_MIC_VOL); + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN | + DA7210_DFLT_MIC_VOL); /* Enable Left and Right input PGA */ snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); @@ -533,6 +581,19 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 and set default vol */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_DFLT_AUX1_VOL | + DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_DFLT_AUX1_VOL | + DA7210_AUX1_R_EN); + /* Enable Aux2 and set default vol */ + snd_soc_write(codec, DA7210_AUX2, DA7210_DFLT_AUX2_VOL | + DA7210_AUX2_EN); + + /* Set default In PGA gain */ + snd_soc_write(codec, DA7210_IN_GAIN, DA7210_DFLT_INPGA_VOL | + (DA7210_DFLT_INPGA_VOL << 4)); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 2

[alsa-devel] [PATCH 1/9] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 13 Oct '11

13 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 55 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 55 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a9d9d39..eef861d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -40,6 +40,9 @@ #define DA7210_DAC_SEL 0x17 #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -133,6 +136,22 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + +/* Default gain/vol values */ +#define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ +#define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ +#define DA7210_DFLT_OUT2_VOL 0x06 /* 0dB */ + #define DA7210_VERSION "0.0.1" /* @@ -150,12 +169,35 @@ * -> min = -79.5 dB */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); + +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("DAC Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), }; /* Codec private data */ @@ -478,6 +520,19 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Set default DAC gain */ + snd_soc_write(codec, DA7210_DAC_L, DA7210_DFLT_DAC_GAIN); + snd_soc_write(codec, DA7210_DAC_R, DA7210_DFLT_DAC_GAIN); + + /* Enable Line out amplifiers and set default vol */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_DFLT_OUT1_VOL | + DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_DFLT_OUT1_VOL | + DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_DFLT_OUT2_VOL | + DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | + DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 2

[alsa-devel] [smatch stuff] reading beyond the end of the array in build_feature_ctl()
by Dan Carpenter 13 Oct '11

13 Oct '11

Hi Daniel, Smatch complains about a potential read past the end of the array in build_feature_ctl(). sound/usb/mixer.c +1035 build_feature_ctl(34) error: buffer overflow 'audio_feature_info' 13 <= 14 This is in the init code so probably it's correct, and I've just missed how it works. Can you take a look? The problem is that audio_feature_info[] has 13 elements, but Smatch thinks control - 1 can go up to 14 so we're two elements past the end. 1035 cval->val_type = audio_feature_info[control-1].type; Smatch get the value for "control" from "i" in parse_audio_feature_unit(). sound/usb/mixer.c 1261 } else { /* UAC_VERSION_2 */ 1262 for (i = 0; i < 30/2; i++) { 1263 unsigned int ch_bits = 0; 1264 unsigned int ch_read_only = 0; 1265 1266 for (j = 0; j < channels; j++) { 1267 unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); 1268 if (uac2_control_is_readable(mask, i)) { 1269 ch_bits |= (1 << j); 1270 if (!uac2_control_is_writeable(mask, i)) 1271 ch_read_only |= (1 << j); 1272 } 1273 } 1274 1275 /* NOTE: build_feature_ctl() will mark the control read-only if all channels 1276 * are marked read-only in the descriptors. Otherwise, the control will be 1277 * reported as writeable, but the driver will not actually issue a write 1278 * command for read-only channels */ 1279 if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ 1280 build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only); ^ "i" can be 13 here. 1281 if (uac2_control_is_readable(master_bits, i)) 1282 build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, ^ "i" maybe can go up to 14 here, I'm not sure. 1283 !uac2_control_is_writeable(master_bits, i)); 1284 } 1285 } regards, dan carpenter

2 1

[alsa-devel] [PATCH] ALSA: HDA: Fixup Realtek headphone pin initialization
by David Henningsson 13 Oct '11

13 Oct '11

This typo caused headphone pins not to be initialized correctly. BugLink: https://bugs.launchpad.net/bugs/871582 Reported-by: Effenberg Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf53663..6a4128d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3263,7 +3263,7 @@ static void alc_auto_init_extra_out(struct hda_codec *codec) int i; hda_nid_t pin, dac; - for (i = 0; i < spec->autocfg.speaker_outs; i++) { + for (i = 0; i < spec->autocfg.hp_outs; i++) { pin = spec->autocfg.hp_pins[i]; if (!pin) break; -- 1.7.5.4

2 1

Re: [alsa-devel] [PATCH] ASoC: da7210: convert to soc-cache
by Ashish Chavan 13 Oct '11

13 Oct '11

Hi Axel, Today I was about to submit a patch series for da7210 codec driver and noticed your patch for the same driver. My patch series is mainly feature update kind of nature. It has total nine patches that add support for various things, starting from basic IOs to DAPM. I am thinking of going forward with my posting which is based on current version of da7210 mainline driver. In am ready to take the responsibility of updating(to base it on my patch series) and testing your patch on the hardware that we have. Other option is to wait till your changes gets approved and then base my patch series on updated driver with your changes. But as this may take some time and I definitely want to get more eyes on the patches early, this would be least preferred. Thanks, -- Ashish (GNU FAN) > Date: Wed, 12 Oct 2011 07:16:25 +0800 > From: Axel Lin <axel.lin(a)gmail.com> > Subject: [alsa-devel] [PATCH] ASoC: da7210: convert to soc-cache > To: linux-kernel(a)vger.kernel.org > Cc: David Chen <Dajun.chen(a)diasemi.com>, alsa-devel(a)alsa-project.org, > Mark Brown <broonie(a)opensource.wolfsonmicro.com>, Liam Girdwood > <lrg(a)ti.com>, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Message-ID: <1318374985.6115.10.camel@phoenix> > Content-Type: text/plain; charset="UTF-8" > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > Hi Kuninori, > This patch converts da7210 to use soc-cache APIs. > I don't have this hardware, I'd appreciate if you can test this patch. > > Thanks, > Axel > sound/soc/codecs/da7210.c | 123 ++++++++++++++++---------------------------- > 1 files changed, 45 insertions(+), 78 deletions(-) > > diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c > index 33b4823..1f30ef9 100644 > --- a/sound/soc/codecs/da7210.c > +++ b/sound/soc/codecs/da7210.c > @@ -162,7 +162,6 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { > /* Codec private data */ > struct da7210_priv { > enum snd_soc_control_type control_type; > - void *control_data; > }; > > /* > @@ -189,50 +188,16 @@ static const u8 da7210_reg[] = { > 0x00, /* R88 */ > }; > > -/* > - * Read da7210 register cache > - */ > -static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) > -{ > - u8 *cache = codec->reg_cache; > - BUG_ON(reg >= ARRAY_SIZE(da7210_reg)); > - return cache[reg]; > -} > - > -/* > - * Write to the da7210 register space > - */ > -static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) > -{ > - u8 *cache = codec->reg_cache; > - u8 data[2]; > - > - BUG_ON(codec->driver->volatile_register); > - > - data[0] = reg & 0xff; > - data[1] = value & 0xff; > - > - if (reg >= codec->driver->reg_cache_size) > - return -EIO; > - > - if (2 != codec->hw_write(codec->control_data, data, 2)) > - return -EIO; > - > - cache[reg] = value; > - return 0; > -} > - > -/* > - * Read from the da7210 register space. > - */ > -static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) > +static int da7210_volatile_register(struct snd_soc_codec *codec, > + unsigned int reg) > { > - if (DA7210_STATUS == reg) > - return i2c_smbus_read_byte_data(codec->control_data, reg); > - > - return da7210_read_reg_cache(codec, reg); > + switch (reg) { > + case DA7210_STATUS: > + return 1; > + default: > + return 0; > + } > } > - > static int da7210_startup(struct snd_pcm_substream *substream, > struct snd_soc_dai *dai) > { > @@ -271,13 +236,13 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, > u32 fs, bypass; > > /* set DAI source to Left and Right ADC */ > - da7210_write(codec, DA7210_DAI_SRC_SEL, > + snd_soc_write(codec, DA7210_DAI_SRC_SEL, > DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); > > /* Enable DAI */ > - da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); > + snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); > > - dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); > + dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); > > switch (params_format(params)) { > case SNDRV_PCM_FORMAT_S16_LE: > @@ -290,7 +255,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, > return -EINVAL; > } > > - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); > + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); > > hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? > DA7210_DAC_HPF : DA7210_ADC_HPF; > @@ -383,8 +348,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) > u32 dai_cfg1; > u32 dai_cfg3; > > - dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); > - dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); > + dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); > + dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); > > switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > case SND_SOC_DAIFMT_CBM_CFM: > @@ -412,8 +377,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) > */ > dai_cfg1 |= DA7210_DAI_FLEN_64BIT; > > - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); > - da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); > + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); > + snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); > > return 0; > } > @@ -452,11 +417,15 @@ static struct snd_soc_dai_driver da7210_dai = { > static int da7210_probe(struct snd_soc_codec *codec) > { > struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); > + int ret; > > dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); > > - codec->control_data = da7210->control_data; > - codec->hw_write = (hw_write_t)i2c_master_send; > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); > + if (ret < 0) { > + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > + return ret; > + } > > /* FIXME > * > @@ -473,8 +442,8 @@ static int da7210_probe(struct snd_soc_codec *codec) > /* > * make sure that DA7210 use bypass mode before start up > */ > - da7210_write(codec, DA7210_STARTUP1, 0); > - da7210_write(codec, DA7210_PLL_DIV3, > + snd_soc_write(codec, DA7210_STARTUP1, 0); > + snd_soc_write(codec, DA7210_PLL_DIV3, > DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); > > /* > @@ -482,36 +451,36 @@ static int da7210_probe(struct snd_soc_codec *codec) > */ > > /* Enable Left & Right MIC PGA and Mic Bias */ > - da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); > - da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); > + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); > + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); > > /* Enable Left and Right input PGA */ > - da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); > - da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); > + snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); > + snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); > > /* Enable Left and Right ADC */ > - da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); > + snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); > > /* > * DAC settings > */ > > /* Enable Left and Right DAC */ > - da7210_write(codec, DA7210_DAC_SEL, > + snd_soc_write(codec, DA7210_DAC_SEL, > DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | > DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); > > /* Enable Left and Right out PGA */ > - da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); > - da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); > + snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); > + snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); > > /* Enable Left and Right HeadPhone PGA */ > - da7210_write(codec, DA7210_HP_CFG, > + snd_soc_write(codec, DA7210_HP_CFG, > DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | > DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); > > /* Diable PLL and bypass it */ > - da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); > + snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); > > /* > * If 48kHz sound came, it use bypass mode, > @@ -522,22 +491,22 @@ static int da7210_probe(struct snd_soc_codec *codec) > * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. > * see da7210_hw_params > */ > - da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ > - da7210_write(codec, DA7210_PLL_DIV2, 0x99); > - da7210_write(codec, DA7210_PLL_DIV3, 0x0A | > + snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ > + snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); > + snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | > DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); > snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); > > /* As suggested by Dialog */ > - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ > - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); > - da7210_write(codec, DA7210_A_PLL1, 0x01); > - da7210_write(codec, DA7210_A_CP_MODE, 0x7C); > - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ > - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); > + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ > + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); > + snd_soc_write(codec, DA7210_A_PLL1, 0x01); > + snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); > + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ > + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); > > /* Activate all enabled subsystem */ > - da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); > + snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); > > snd_soc_add_controls(codec, da7210_snd_controls, > ARRAY_SIZE(da7210_snd_controls)); > @@ -549,11 +518,10 @@ static int da7210_probe(struct snd_soc_codec *codec) > > static struct snd_soc_codec_driver soc_codec_dev_da7210 = { > .probe = da7210_probe, > - .read = da7210_read, > - .write = da7210_write, > .reg_cache_size = ARRAY_SIZE(da7210_reg), > .reg_word_size = sizeof(u8), > .reg_cache_default = da7210_reg, > + .volatile_register = da7210_volatile_register, > }; > > #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) > @@ -568,7 +536,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, > return -ENOMEM; > > i2c_set_clientdata(i2c, da7210); > - da7210->control_data = i2c; > da7210->control_type = SND_SOC_I2C; > > ret = snd_soc_register_codec(&i2c->dev, > -- > 1.7.4.1 >

3 5

[alsa-devel] [PATCH] [RESEND] Aplay: Fix spelling error
by David Henningsson 12 Oct '11

12 Oct '11

BugLink: https://bugs.launchpad.net/bugs/816388 Reported-by: Chad Dunlap <cldunlap1(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Resending patch to more maintainers, as nobody seem to have picked it up the previous time. aplay/aplay.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 07e6bc0..5ab63a9 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -209,7 +209,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --disable-format disable automatic format conversions\n" " --disable-softvol disable software volume control (softvol)\n" " --test-position test ring buffer position\n" -" --test-coef=# test coeficient for ring buffer position (default 8)\n" +" --test-coef=# test coefficient for ring buffer position (default 8)\n" " expression for validation is: coef * (buffer_size / 2)\n" " --test-nowait do not wait for ring buffer - eats whole CPU\n" " --max-file-time=# start another output file when the old file has recorded\n" -- 1.7.5.4

2 1

[alsa-devel] SOC: using a codec (ADAU1761) just as an analog mixer
by Matteo Fortini 12 Oct '11

12 Oct '11

I'm using an ADAU1761 codec in a project, but we'd like, for now, to use it just as a mixer, i.e. without passing through the ADC/DAC and DSP stages. Which is the best way of accomplishing that? In the past I tried to add virtual DAPM routes in order to get the relevant power systems powered up, and bypassing the standby modes, since the device wouldn't let me use the mixer controls through I2C if I don't clock it. Another question: how do I tell ALSA to power up a stage (in this case, the ALC/differential input stage)? TIA, Matteo

2 1

[alsa-devel] [PATCH] ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X
by Axel Lin 12 Oct '11

12 Oct '11

In saarb_pm860x_init() and evb3_pm860x_init(), we call pm860x_hs_jack_detect() and pm860x_mic_jack_detect() which in turn calls pm860x_set_bits(). Thus make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X. This patch fixes below build error if CONFIG_MFD_88PM860X is not configured. LD .tmp_vmlinux1 sound/built-in.o: In function `pm860x_write_reg_cache': last.c:(.text+0x29e9c): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_set_bias_level': last.c:(.text+0x29ecc): undefined reference to `pm860x_set_bits' last.c:(.text+0x29f00): undefined reference to `pm860x_reg_write' last.c:(.text+0x29f18): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_read_reg_cache': last.c:(.text+0x29f40): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_probe': last.c:(.text+0x2a034): undefined reference to `pm860x_bulk_read' sound/built-in.o: In function `pm860x_codec_handler': last.c:(.text+0x2a344): undefined reference to `pm860x_reg_read' last.c:(.text+0x2a354): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_mic_jack_detect': last.c:(.text+0x2a450): undefined reference to `pm860x_set_bits' sound/built-in.o: In function `pm860x_hs_jack_detect': last.c:(.text+0x2a4d0): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a4f8): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a510): undefined reference to `pm860x_set_bits' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/Kconfig | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46b..ffd2242 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -121,6 +121,7 @@ config SND_PXA2XX_SOC_PALM27X config SND_SOC_SAARB tristate "SoC Audio support for Marvell Saarb" depends on SND_PXA2XX_SOC && MACH_SAARB + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help @@ -130,6 +131,7 @@ config SND_SOC_SAARB config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help -- 1.7.4.1

2 1

[alsa-devel] [PATCH 9/9] ASoC: da7210: Add support for DAPM
by Ashish Chavan 12 Oct '11

12 Oct '11

This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 251 +++++++++++++++++++++++++++++++++++++++++---- 1 files changed, 231 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 2c6e47c..e0c3c5b 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -29,6 +29,8 @@ #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_AUX1_L 0x09 @@ -199,6 +201,24 @@ #define DA7210_HP_L_ZC (1 << 6) #define DA7210_HP_R_ZC (1 << 7) +/* STARTUP2 bit fields */ +#define DA7210_LOUT1_L_STBY (1 << 0) +#define DA7210_LOUT1_R_STBY (1 << 1) +#define DA7210_LOUT2_STBY (1 << 2) +#define DA7210_HP_L_STBY (1 << 3) +#define DA7210_HP_R_STBY (1 << 4) +#define DA7210_DAC_L_STBY (1 << 5) +#define DA7210_DAC_R_STBY (1 << 6) + +/* STARTUP3 bit fields */ +#define DA7210_MIC_L_STBY (1 << 0) +#define DA7210_MIC_R_STBY (1 << 1) +#define DA7210_LIN1_L_STBY (1 << 2) +#define DA7210_LIN1_R_STBY (1 << 3) +#define DA7210_LIN2_STBY (1 << 4) +#define DA7210_ADC_L_STBY (1 << 5) +#define DA7210_ADC_R_STBY (1 << 6) + /* Default gain/vol values */ #define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ #define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ @@ -460,6 +480,200 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), }; +/* + * DAPM Controls + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_MICBIAS("Mic Bias", DA7210_MIC_L, 6, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Bias", NULL, "MICR"}, + {"Mic Bias", NULL, "MICL"}, + + {"Mic Left", NULL, "Mic Bias"}, + {"Mic Right", NULL, "Mic Bias"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, +}; + +static int da7210_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, da7210_dapm_widgets, + ARRAY_SIZE(da7210_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, da7210_audio_map, + ARRAY_SIZE(da7210_audio_map)); + + return 0; +} + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -499,25 +713,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Enable Mic,AUX1 and AUX2 */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD); - } - - return 0; -} /* * Set PCM DAI word length. @@ -675,7 +870,6 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, }; @@ -748,6 +942,10 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + /* In mixer source selection, only MIC is selected by default */ + snd_soc_write(codec, DA7210_INMIX_L, 0x01); + snd_soc_write(codec, DA7210_INMIX_R, 0x01); + /* Enable Left and Right ADC */ snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); @@ -764,6 +962,10 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + /* Outmixer source selection, only DAC is selected by default */ + snd_soc_write(codec, DA7210_OUTMIX_L, 0x10); + snd_soc_write(codec, DA7210_OUTMIX_R, 0x10); + /* Enable Left and Right HeadPhone PGA */ snd_soc_write(codec, DA7210_HP_CFG, DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | @@ -797,6 +999,14 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable ramp mode for DAC gain update */ snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Enable standby for all inputs and outputs */ + snd_soc_write(codec, DA7210_STARTUP2, DA7210_LOUT1_L_STBY | + DA7210_LOUT1_R_STBY | DA7210_LOUT2_STBY | DA7210_HP_L_STBY | + DA7210_HP_R_STBY | DA7210_DAC_L_STBY | DA7210_DAC_R_STBY); + snd_soc_write(codec, DA7210_STARTUP3, DA7210_MIC_L_STBY | + DA7210_MIC_R_STBY | DA7210_LIN1_L_STBY | DA7210_LIN1_R_STBY | + DA7210_LIN2_STBY | DA7210_ADC_L_STBY | DA7210_ADC_R_STBY); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); @@ -828,6 +1038,7 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, da7210_snd_controls, ARRAY_SIZE(da7210_snd_controls)); + da7210_add_widgets(codec); dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); -- 1.7.1

2 1

[alsa-devel] [PATCH 8/9] ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC
by Ashish Chavan 12 Oct '11

12 Oct '11

This patch add controls for setting cut-off for high pass and voice filters of ADC and DAC. There are also switches to enable/disable these filters. Also removed hard coded, fixed values of these parameters used by previous version of driver. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 57 +++++++++++++++++++++++++------------------- 1 files changed, 32 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 9cab62a..2c6e47c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -274,6 +274,28 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -426,6 +448,16 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), }; /* Codec private data */ @@ -497,7 +529,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ @@ -528,68 +559,45 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; - switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -599,7 +607,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); -- 1.7.1

2 1

[alsa-devel] [PATCH 5/9] ASoC: da7210: Add support for other DAI word lengths, format and mode
by Ashish Chavan 12 Oct '11

12 Oct '11

This patchs adds support for following, (1) DAI 20 and 32 bit word sizes (2) DAI left and right justified formats (3) DAI slave mode Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 23 ++++++++++++++++++++++- 1 files changed, 22 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 1b4554a..60f403b 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -120,12 +120,17 @@ /* DAI_CFG1 bit fields */ #define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S20_3LE (1 << 0) #define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_WORD_S32_LE (3 << 0) #define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_SLAVE (0 << 7) #define DA7210_DAI_MODE_MASTER (1 << 7) /* DAI_CFG3 bit fields */ #define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_FORMAT_LEFT_J (1 << 0) +#define DA7210_DAI_FORMAT_RIGHT_J (2 << 0) #define DA7210_DAI_OE (1 << 3) #define DA7210_DAI_EN (1 << 7) @@ -401,9 +406,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; + break; case SNDRV_PCM_FORMAT_S24_LE: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_cfg1 |= DA7210_DAI_WORD_S32_LE; + break; default: return -EINVAL; } @@ -508,6 +519,9 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_CBM_CFM: dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_cfg1 |= DA7210_DAI_MODE_SLAVE; + break; default: return -EINVAL; } @@ -520,6 +534,12 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_I2S: dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; break; + case SND_SOC_DAIFMT_LEFT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J; + break; default: return -EINVAL; } @@ -536,7 +556,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } -#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { -- 1.7.1

2 1

[alsa-devel] [PATCH 4/9] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 12 Oct '11

12 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX, ADC and DAC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index b48a8af..1b4554a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -45,6 +45,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -56,6 +57,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -166,6 +168,9 @@ /* AUX2 bit fields */ #define DA7210_AUX2_EN (1 << 3) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + /* Default gain/vol values */ #define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ #define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ @@ -227,6 +232,13 @@ static const char *da7210_mic_bias_voltage_txt[] = { static const struct soc_enum da7210_mic_bias = SOC_ENUM_SINGLE(DA7210_MIC_L, 4, 4, da7210_mic_bias_voltage_txt); +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -284,6 +296,24 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), + + /* Mute controls */ + SOC_SINGLE("Mic Left Mute Switch", DA7210_MIC_L, 3, 1, 0), + SOC_SINGLE("Mic Right Mute Switch", DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Mute Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Mute Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Mute Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("DAC Mute Switch", DA7210_DAC_HPF, 2, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("DAC Soft Mute Rate", DA7210_SOFTMUTE, 0, 0X7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -629,6 +659,8 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Set default In PGA gain */ snd_soc_write(codec, DA7210_IN_GAIN, DA7210_DFLT_INPGA_VOL | (DA7210_DFLT_INPGA_VOL << 4)); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] [PATCH 00/09] ASoC: da7210: Add support for all IOs and other features
by Ashish Chavan 12 Oct '11

12 Oct '11

Hi, Existing da7210 driver is very minimalistic in terms of functionality. It supports only one output i.e. head phone. da7210 codec hardware supports many more IOs and features. Here is a patch series that is "feature update" in nature. This adds support for all inputs(MIC L-R, AUX1 L-R, AUX2) and outputs (OUT1 L-R, OUT2) as well as following features, - ADC and DAC equalizers - Mute - Zero cross - Other DAI word lengths, format and mode - ALC - Noise suppression - High pass and voice filters - DAPM This series also contains a small bug fix for head phone volume control. Thanks, -- Ashish (GNU FAN) --- Ashish Chavan (9): ASoC: da7210: Add support for line out and DAC ASoC: da7210: Add support for line input and mic ASoC: da7210: Add support for ADC & DAC equalizers ASoC: da7210: Add support for mute and zero cross controls ASoC: da7210: Add support for other DAI length, format and mode ASoC: da7210: bugfix for head phone volume control ASoC: da7210: Add support for ALC and Noise suppression ASoC: da7210: Add support for High pass and Voice filters ASoC: da7210: Add support for DAPM sound/soc/codecs/da7210.c | 642 +++++++++++++++++++++++++++++++++++--- 1 files changed, 586 insertions(+), 56 deletions(-) -- 1.7.1

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[alsa-devel] [PATCH v2] ASoC: twl6040: Workaround for headset DC offset caused pop noise
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Both Headset DAC need to be turned on/off at the same time before any of the output drivers are enabled (HS Left/Right, Earpiece). Move the HS DAC enable code to sequenced DAPM_SUPPLY, and attach it to the DACs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Changes since v1: - Use DAPM_SUPPLY_S for the HS DAC enable code Regards, Peter include/linux/mfd/twl6040.h | 1 + sound/soc/codecs/twl6040.c | 36 ++++++++++++++++++++++++++++-------- 2 files changed, 29 insertions(+), 8 deletions(-) diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index 87a4778..2463c261 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -122,6 +122,7 @@ /* HSLCTL/R (0x10/0x11) fields */ +#define TWL6040_HSDACENA (1 << 0) #define TWL6040_HSDACMODE (1 << 1) #define TWL6040_HSDRVMODE (1 << 3) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 8648498..6369230 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -654,6 +654,26 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + u8 hslctl, hsrctl; + + /* + * Workaround for Headset DC offset caused pop noise: + * Both HS DAC need to be turned on (before the HS driver) and off at + * the same time. + */ + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + if (SND_SOC_DAPM_EVENT_ON(event)) { + hslctl |= TWL6040_HSDACENA; + hsrctl |= TWL6040_HSDACENA; + } else { + hslctl &= ~TWL6040_HSDACENA; + hsrctl &= ~TWL6040_HSDACENA; + } + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + msleep(1); return 0; } @@ -1103,14 +1123,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0, - twl6040_hs_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -1175,6 +1189,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { NULL, 0), SND_SOC_DAPM_SUPPLY("Vibra Right Control", TWL6040_REG_VIBCTLR, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("HSDAC Power", 1, SND_SOC_NOPM, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HF Left PGA", @@ -1204,6 +1221,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"AFMAmpL", NULL, "AFML"}, {"AFMAmpR", NULL, "AFMR"}, + {"HSDAC Left", NULL, "HSDAC Power"}, + {"HSDAC Right", NULL, "HSDAC Power"}, + {"Headset Left Playback", "HS DAC", "HSDAC Left"}, {"Headset Left Playback", "Line-In amp", "AFMAmpL"}, -- 1.7.7

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[alsa-devel] [PATCH] ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 depend on MFD_88PM860X
by Axel Lin 12 Oct '11

12 Oct '11

In saarb_pm860x_init() and evb3_pm860x_init(), we call pm860x_hs_jack_detect() and pm860x_mic_jack_detect() which in turn calls pm860x_set_bits(). Thus make SND_SOC_SAARB and SND_SOC_TAVOREVB3 depend on MFD_88PM860X This patch fixes below build error if CONFIG_MFD_88PM860X is not configured. LD .tmp_vmlinux1 sound/built-in.o: In function `pm860x_write_reg_cache': last.c:(.text+0x29e9c): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_set_bias_level': last.c:(.text+0x29ecc): undefined reference to `pm860x_set_bits' last.c:(.text+0x29f00): undefined reference to `pm860x_reg_write' last.c:(.text+0x29f18): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_read_reg_cache': last.c:(.text+0x29f40): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_probe': last.c:(.text+0x2a034): undefined reference to `pm860x_bulk_read' sound/built-in.o: In function `pm860x_codec_handler': last.c:(.text+0x2a344): undefined reference to `pm860x_reg_read' last.c:(.text+0x2a354): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_mic_jack_detect': last.c:(.text+0x2a450): undefined reference to `pm860x_set_bits' sound/built-in.o: In function `pm860x_hs_jack_detect': last.c:(.text+0x2a4d0): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a4f8): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a510): undefined reference to `pm860x_set_bits' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/Kconfig | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46b..b12b4af 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -121,6 +121,7 @@ config SND_PXA2XX_SOC_PALM27X config SND_SOC_SAARB tristate "SoC Audio support for Marvell Saarb" depends on SND_PXA2XX_SOC && MACH_SAARB + depends on MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help @@ -130,6 +131,7 @@ config SND_SOC_SAARB config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + depends on MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help -- 1.7.4.1

2 1

[alsa-devel] [PATCH v3 0/5] ASoC/MFD/Input: twl6040: Support for audio driven vibra
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Hello, Changes since v2: - Rebased on git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch - Acked-by lines added to patches. Intro mail form v1: The following series adds support for the soc driver to route audio (PCM stream) to the vibra ports of the twl6040. We have ALSA controls for the vibra source selection (PDM, or Input/FF). The route can not be changed from Input/FF to PDM during ongoing effect coming from Input/FF. The Input/FF driver can not execute new effect, while the vibra source is configured to be PDM. TO get things working in legacy mode the omap-mcpdm, and twl6040 soc codec driver's channel count needed to be fixed. To minimize the need for reads through the I2C bus, the MFD driver will cache the vibra control registers. With this series we can route PCM audio to drive the twl6040 vibra output. In order to that we need to play 5 channel audio, and place the vibra sample at the 5th channel. This will change with ABE support (change == simplified). The series has been created on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch If there are no objections, it would be good if this series goes via audio. Regards, Peter --- Peter Ujfalusi (5): Input: twl6040: Simplify vibra regsiter definitions MFD: twl6040: Cache the vibra control registers MFD: twl6040: function to query the vibra status for clients Input: twl6040-vibra: Check the selected path for vibra ASoC: twl6040: Support for vibra output paths drivers/input/misc/twl6040-vibra.c | 19 ++++++--- drivers/mfd/twl6040-core.c | 31 +++++++++++++-- include/linux/mfd/twl6040.h | 24 +++++------ sound/soc/codecs/twl6040.c | 72 ++++++++++++++++++++++++++++++++++++ 4 files changed, 123 insertions(+), 23 deletions(-) -- 1.7.7

2 6

[alsa-devel] [PATCH] ASoC: twl6040: Workaround for headset DC offset caused pop noise
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Both Headset DAC need to be enabled at the same time, before any of the output drivers are enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- include/linux/mfd/twl6040.h | 1 + sound/soc/codecs/twl6040.c | 42 ++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 41 insertions(+), 2 deletions(-) diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index d9e05ea..0bbb1e7 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -122,6 +122,7 @@ /* HSLCTL/R (0x10/0x11) fields */ +#define TWL6040_HSDACENA (1 << 0) #define TWL6040_HSDACMODE (1 << 1) #define TWL6040_HSDRVMODE (1 << 3) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 93f8a59..924eaf3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -91,6 +91,7 @@ struct twl6040_data { int pll_power_mode; int hs_power_mode; int hs_power_mode_locked; + int hs_dac_enabled; unsigned int clk_in; unsigned int sysclk; u16 hs_left_step; @@ -654,6 +655,43 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + u8 hslctl, hsrctl; + + /* + * Workaround for Headset DC offset caused pop noise: + * Both HS DAC need to be turned on (before the HS driver) and off at + * the same time. + */ + if (SND_SOC_DAPM_EVENT_ON(event)) { + if (!priv->hs_dac_enabled++) { + hslctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSLCTL); + hslctl |= TWL6040_HSDACENA; + + hsrctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSRCTL); + hsrctl |= TWL6040_HSDACENA; + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + } + } else { + if (!--priv->hs_dac_enabled) { + hslctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSLCTL); + hslctl &= ~TWL6040_HSDACENA; + + hsrctl = twl6040_read_reg_cache(codec, + TWL6040_REG_HSRCTL); + hsrctl &= ~TWL6040_HSDACENA; + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + } + } + msleep(1); return 0; } @@ -1061,11 +1099,11 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0, + SND_SOC_NOPM, 0, 0, twl6040_hs_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0, + SND_SOC_NOPM, 0, 0, twl6040_hs_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", -- 1.7.7

2 6

[alsa-devel] [PATCH] ASoC: da7210: convert to soc-cache
by Axel Lin 12 Oct '11

12 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi Kuninori, This patch converts da7210 to use soc-cache APIs. I don't have this hardware, I'd appreciate if you can test this patch. Thanks, Axel sound/soc/codecs/da7210.c | 123 ++++++++++++++++---------------------------- 1 files changed, 45 insertions(+), 78 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 33b4823..1f30ef9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -162,7 +162,6 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -189,50 +188,16 @@ static const u8 da7210_reg[] = { 0x00, /* R88 */ }; -/* - * Read da7210 register cache - */ -static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) -{ - u8 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(da7210_reg)); - return cache[reg]; -} - -/* - * Write to the da7210 register space - */ -static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) -{ - u8 *cache = codec->reg_cache; - u8 data[2]; - - BUG_ON(codec->driver->volatile_register); - - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (reg >= codec->driver->reg_cache_size) - return -EIO; - - if (2 != codec->hw_write(codec->control_data, data, 2)) - return -EIO; - - cache[reg] = value; - return 0; -} - -/* - * Read from the da7210 register space. - */ -static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) +static int da7210_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { - if (DA7210_STATUS == reg) - return i2c_smbus_read_byte_data(codec->control_data, reg); - - return da7210_read_reg_cache(codec, reg); + switch (reg) { + case DA7210_STATUS: + return 1; + default: + return 0; + } } - static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -271,13 +236,13 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, u32 fs, bypass; /* set DAI source to Left and Right ADC */ - da7210_write(codec, DA7210_DAI_SRC_SEL, + snd_soc_write(codec, DA7210_DAI_SRC_SEL, DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); /* Enable DAI */ - da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); - dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -290,7 +255,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? DA7210_DAC_HPF : DA7210_ADC_HPF; @@ -383,8 +348,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) u32 dai_cfg1; u32 dai_cfg3; - dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); - dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -412,8 +377,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) */ dai_cfg1 |= DA7210_DAI_FLEN_64BIT; - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); return 0; } @@ -452,11 +417,15 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* FIXME * @@ -473,8 +442,8 @@ static int da7210_probe(struct snd_soc_codec *codec) /* * make sure that DA7210 use bypass mode before start up */ - da7210_write(codec, DA7210_STARTUP1, 0); - da7210_write(codec, DA7210_PLL_DIV3, + snd_soc_write(codec, DA7210_STARTUP1, 0); + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* @@ -482,36 +451,36 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* Enable Left & Right MIC PGA and Mic Bias */ - da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); - da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); /* Enable Left and Right input PGA */ - da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); - da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); /* Enable Left and Right ADC */ - da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); /* * DAC settings */ /* Enable Left and Right DAC */ - da7210_write(codec, DA7210_DAC_SEL, + snd_soc_write(codec, DA7210_DAC_SEL, DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); /* Enable Left and Right out PGA */ - da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); - da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); /* Enable Left and Right HeadPhone PGA */ - da7210_write(codec, DA7210_HP_CFG, + snd_soc_write(codec, DA7210_HP_CFG, DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); /* Diable PLL and bypass it */ - da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); /* * If 48kHz sound came, it use bypass mode, @@ -522,22 +491,22 @@ static int da7210_probe(struct snd_soc_codec *codec) * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. * see da7210_hw_params */ - da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - da7210_write(codec, DA7210_PLL_DIV2, 0x99); - da7210_write(codec, DA7210_PLL_DIV3, 0x0A | + snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); + snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); - da7210_write(codec, DA7210_A_PLL1, 0x01); - da7210_write(codec, DA7210_A_CP_MODE, 0x7C); - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + snd_soc_write(codec, DA7210_A_PLL1, 0x01); + snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ - da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); snd_soc_add_controls(codec, da7210_snd_controls, ARRAY_SIZE(da7210_snd_controls)); @@ -549,11 +518,10 @@ static int da7210_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, - .read = da7210_read, - .write = da7210_write, .reg_cache_size = ARRAY_SIZE(da7210_reg), .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, + .volatile_register = da7210_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -568,7 +536,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, da7210); - da7210->control_data = i2c; da7210->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2] ASoC: Remove impossible case from wm8994_hw_params
by Axel Lin 12 Oct '11

12 Oct '11

We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params. Thus no need to check wm8994-aif3 in wm8994_hw_params. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: original patch does not apply after commit 81204c84c "ASoC: Add WM1811 support". sound/soc/codecs/wm8994.c | 10 ---------- 1 files changed, 0 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 16542de..68e769e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2235,7 +2235,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; int aif2_reg; @@ -2278,15 +2277,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "AIF2 using split LRCLK\n"); } break; - case 3: - switch (control->type) { - case WM1811: - case WM8958: - aif1_reg = WM8958_AIF3_CONTROL_1; - break; - default: - return 0; - } default: return -EINVAL; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Delete ads117x.h
by Axel Lin 12 Oct '11

12 Oct '11

This is not required after multi-component patch. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ads117x.h | 13 ------------- 1 files changed, 0 insertions(+), 13 deletions(-) delete mode 100644 sound/soc/codecs/ads117x.h diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h deleted file mode 100644 index 3ce0286..0000000 --- a/sound/soc/codecs/ads117x.h +++ /dev/null @@ -1,13 +0,0 @@ -/* - * ads117x.h -- Driver for ads1174/8 ADC chips - * - * Copyright 2009 ShotSpotter Inc. - * Author: Graeme Gregory <gg(a)slimlogic.co.uk> - * - * 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. - */ -extern struct snd_soc_dai_driver ads117x_dai; -extern struct snd_soc_codec_driver soc_codec_dev_ads117x; -- 1.7.4.1

2 1

[alsa-devel] Driver for daisy-chained AD1938 audio codecs?
by Steve Strobel 12 Oct '11

12 Oct '11

I am considering a Blackfin DSP hardware design which would need eight channels of audio input/output. I haven't been able to find a single audio codec with more than six input channels, so I am looking at daisy-chaining AD1938 codecs as described on page 19 of its datasheet [1]. In the past I worked on a prototype that used two AD1938 chips on separate SPORT interfaces. It worked, but having to deal with two ALSA sound devices made the software relatively complicated (error recovery, synchronization, etc). It would be nice if all of the channels appeared as a single ALSA device. I believe that daisy-chained AD1938 chips would work that way, but I haven't found an ALSA driver for that configuration. Does such a driver already exist? Any suggestions for making one? Thanks for any input. Steve [1] - http://www.analog.com/static/imported-files/data_sheets/AD1938.pdf -- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

2 1

[alsa-devel] [PATCH v2 0/6] ASoC/MFD/Input: twl6040: Support for audio driven vibra
by Peter Ujfalusi 12 Oct '11

12 Oct '11

Hello, Changes since v1: - Patches already taken by Mark has been removed - Patch 4: Return -EBUSY instead of -EPERM Intro mail form v1: The following series adds support for the soc driver to route audio (PCM stream) to the vibra ports of the twl6040. We have ALSA controls for the vibra source selection (PDM, or Input/FF). The route can not be changed from Input/FF to PDM during ongoing effect coming from Input/FF. The Input/FF driver can not execute new effect, while the vibra source is configured to be PDM. TO get things working in legacy mode the omap-mcpdm, and twl6040 soc codec driver's channel count needed to be fixed. To minimize the need for reads through the I2C bus, the MFD driver will cache the vibra control registers. With this series we can route PCM audio to drive the twl6040 vibra output. In order to that we need to play 5 channel audio, and place the vibra sample at the 5th channel. This will change with ABE support (change == simplified). The series has been created on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch If there are no objections, it would be good if this series goes via audio. Regards, Peter --- Peter Ujfalusi (6): Input: twl6040: Simplify vibra regsiter definitions MFD: twl6040: Cache the vibra control registers MFD: twl6040: function to query the vibra status for clients Input: twl6040-vibra: Check the selected path for vibra ASoC: omap-mcpdm: Correct the supported number of channels ASoC: twl6040: Support for vibra output paths drivers/input/misc/twl6040-vibra.c | 19 ++++++--- drivers/mfd/twl6040-core.c | 31 +++++++++++++-- include/linux/mfd/twl6040.h | 24 +++++------ sound/soc/codecs/twl6040.c | 72 ++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.c | 14 ++++--- 5 files changed, 131 insertions(+), 29 deletions(-) -- 1.7.6.1

4 12

[alsa-devel] [PATCH] ASoC: tlv320aic23: convert to soc-cache
by Axel Lin 12 Oct '11

12 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, This patch converts tlv320aic23 to use soc-cache APIs. I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/tlv320aic23.c | 169 +++++++++++++--------------------------- 1 files changed, 54 insertions(+), 115 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3a4bb2..c9d85db 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -47,63 +47,6 @@ static const u16 tlv320aic23_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ }; -/* - * read tlv320aic23 register cache - */ -static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec - *codec, unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return -1; - return cache[reg]; -} - -/* - * write tlv320aic23 register cache - */ -static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u16 value) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return; - cache[reg] = value; -} - -/* - * write to the tlv320aic23 register space - */ -static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - - u8 data[2]; - - /* TLV320AIC23 has 7 bit address and 9 bits of data - * so we need to switch one data bit into reg and rest - * of data into val - */ - - if (reg > 9 && reg != 15) { - printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); - return -1; - } - - data[0] = (reg << 1) | (value >> 8 & 0x01); - data[1] = value & 0xff; - - tlv320aic23_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - - printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, - value, reg); - - return -EIO; -} - static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -139,8 +82,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, */ val = (val >= 4) ? 4 : (3 - val); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); - tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); + snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } @@ -151,7 +94,7 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val; - val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; @@ -232,7 +175,6 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { enum snd_soc_control_type control_type; - void *control_data; int mclk; int requested_adc; int requested_dac; @@ -344,7 +286,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int src = snd_soc_read(codec, TLV320AIC23_SRATE); int sr = (src >> 2) & 0x0f; int val = (mclk / bosr_usb_divisor_table[src & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; @@ -368,7 +310,7 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + snd_soc_write(codec, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; @@ -407,9 +349,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - iface_reg = - tlv320aic23_read_reg_cache(codec, - TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -423,7 +364,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg |= (0x03 << 2); break; } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -435,7 +376,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; /* set active */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); return 0; } @@ -450,7 +391,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; @@ -463,14 +404,14 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; u16 reg; - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + reg = snd_soc_read(codec, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; - tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + snd_soc_write(codec, TLV320AIC23_DIGT, reg); return 0; } @@ -481,8 +422,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 iface_reg; - iface_reg = - tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -516,7 +456,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -532,26 +472,26 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ TLV320AIC23_DAC_OFF); - tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + snd_soc_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ - TLV320AIC23_CLK_OFF); + snd_soc_write(codec, TLV320AIC23_PWR, + reg | TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->dapm.bias_level = level; @@ -598,13 +538,7 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec, static int tlv320aic23_resume(struct snd_soc_codec *codec) { - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } + snd_soc_cache_sync(codec); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -613,46 +547,54 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); - int reg; + int ret; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec->control_data = aic23->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; - codec->hw_read = NULL; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* Reset codec */ - tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + snd_soc_write(codec, TLV320AIC23_RESET, 0); + + /* Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + codec->cache_only = 1; + snd_soc_write(codec, 0x0A, 0); + snd_soc_write(codec, 0x0B, 0); + snd_soc_write(codec, 0x0C, 0); + snd_soc_write(codec, 0x0D, 0); + snd_soc_write(codec, 0x0E, 0); + codec->cache_only = 0; /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); - tlv320aic23_write(codec, TLV320AIC23_LINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - (TLV320AIC23_LRS_ENABLED)); + snd_soc_update_bits(codec, TLV320AIC23_LINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); - tlv320aic23_write(codec, TLV320AIC23_RINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - TLV320AIC23_LRS_ENABLED); + snd_soc_update_bits(codec, TLV320AIC23_RINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); - tlv320aic23_write(codec, TLV320AIC23_ANLG, - (reg) & (~TLV320AIC23_BYPASS_ON) & - (~TLV320AIC23_MICM_MUTED)); + snd_soc_update_bits(codec, TLV320AIC23_ANLG, + TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, + 0); /* Default output volume */ - tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); @@ -674,8 +616,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, - .read = tlv320aic23_read_reg_cache, - .write = tlv320aic23_write, .set_bias_level = tlv320aic23_set_bias_level, .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), @@ -702,7 +642,6 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, aic23); - aic23->control_data = i2c; aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

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[alsa-devel] Realtek ALC5630 codec driver
by 卞磊 12 Oct '11

12 Oct '11

Hi ALL, I can't find the Realtek ALC5630 codec driver in current kernel. Where can i get it? Thanks. BRs, Lei.Bian

1 0

[alsa-devel] Driver for daisy-chained AD1938 audio codecs?
by Steve Strobel 11 Oct '11

11 Oct '11

I am considering a Blackfin DSP hardware design which would need eight channels of audio input/output. I haven't been able to find a single audio codec with more than six input channels, so I am looking at daisy-chaining AD1938 codecs as described on page 19 of its datasheet [1]. In the past I worked on a prototype that used two AD1938 chips on separate SPORT interfaces. It worked, but having to deal with two ALSA sound devices made the software relatively complicated (error recovery, synchronization, etc). It would be nice if all of the channels appeared as a single ALSA device. I believe that daisy-chained AD1938 chips would work that way, but I haven't found an ALSA driver for that configuration. Does such a driver already exist? Any suggestions for making one? Thanks for any input. Steve [1] - http://www.analog.com/static/imported-files/data_sheets/AD1938.pdf -- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

1 0

[alsa-devel] [PATCH] ASoC: Implement WM5100 accessory detection support
by Mark Brown 11 Oct '11

11 Oct '11

The WM5100 includes an advanced, low power, accessory detect subsystem capable of detecting both accessory presence and button presses while the device is in an ultra low power mode. Implement initial support for this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 162 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm5100.h | 2 + 2 files changed, 164 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 80f3fee..9ce3e57 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -26,6 +26,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/jack.h> #include <sound/initval.h> #include <sound/tlv.h> #include <sound/wm5100.h> @@ -68,6 +69,11 @@ struct wm5100_priv { bool out_ena[2]; + struct snd_soc_jack *jack; + bool jack_detecting; + bool jack_mic; + int jack_mode; + struct wm5100_fll fll[2]; struct wm5100_pdata pdata; @@ -2114,6 +2120,159 @@ static int wm5100_dig_vu[] = { WM5100_DAC_DIGITAL_VOLUME_6R, }; +static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; + + BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); + + gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); + snd_soc_update_bits(codec, WM5100_ACCESSORY_DETECT_MODE_1, + WM5100_ACCDET_BIAS_SRC_MASK | + WM5100_ACCDET_SRC, + (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | + mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + + wm5100->jack_mode = the_mode; + + dev_dbg(codec->dev, "Set microphone polarity to %d\n", + wm5100->jack_mode); +} + +static void wm5100_micd_irq(struct snd_soc_codec *codec) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + int val; + + val = snd_soc_read(codec, WM5100_MIC_DETECT_3); + + dev_dbg(codec->dev, "Microphone event: %x\n", val); + + if (!(val & WM5100_ACCDET_VALID)) { + dev_warn(codec->dev, "Microphone detection state invalid\n"); + return; + } + + /* No accessory, reset everything and report removal */ + if (!(val & WM5100_ACCDET_STS)) { + dev_dbg(codec->dev, "Jack removal detected\n"); + wm5100->jack_mic = false; + wm5100->jack_detecting = true; + snd_soc_jack_report(wm5100->jack, 0, + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + WM5100_ACCDET_RATE_MASK); + return; + } + + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. + */ + if (val & 0x400) { + if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm5100->jack_mic = true; + snd_soc_jack_report(wm5100->jack, + SND_JACK_HEADSET, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 5 << WM5100_ACCDET_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm5100->jack, 0, SND_JACK_BTN_0); + } + + return; + } + + /* If we detected a lower impedence during initial startup + * then we probably have the wrong polarity, flip it. Don't + * do this for the lowest impedences to speed up detection of + * plain headphones. + */ + if (wm5100->jack_detecting && (val & 0x3f8)) { + wm5100_set_detect_mode(codec, !wm5100->jack_mode); + + return; + } + + /* Don't distinguish between buttons, just report any low + * impedence as BTN_0. + */ + if (val & 0x3fc) { + if (wm5100->jack_mic) { + dev_dbg(codec->dev, "Mic button detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + } else if (wm5100->jack_detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, + SND_JACK_HEADPHONE); + + /* Increase the detection rate a bit for + * responsiveness. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); + } + } +} + +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + + if (jack) { + wm5100->jack = jack; + wm5100->jack_detecting = true; + + wm5100_set_detect_mode(codec, 0); + + /* Slowest detection rate, gives debounce for initial + * detection */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_BIAS_STARTTIME_MASK | + WM5100_ACCDET_RATE_MASK, + (7 << WM5100_ACCDET_BIAS_STARTTIME_SHIFT) | + WM5100_ACCDET_RATE_MASK); + + /* We need the charge pump to power MICBIAS */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_sync(&codec->dapm); + + /* We start off just enabling microphone detection - even a + * plain headphone will trigger detection. + */ + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, WM5100_ACCDET_ENA); + + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_ACCDET_EINT, 0); + } else { + snd_soc_update_bits(codec, WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT, + WM5100_IM_HPDET_EINT | + WM5100_IM_ACCDET_EINT); + snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, + WM5100_ACCDET_ENA, 0); + wm5100->jack = NULL; + } + + return 0; +} + static irqreturn_t wm5100_irq(int irq, void *data) { struct snd_soc_codec *codec = data; @@ -2145,6 +2304,9 @@ static irqreturn_t wm5100_irq(int irq, void *data) complete(&wm5100->fll[1].lock); } + if (irq_val & WM5100_ACCDET_EINT) + wm5100_micd_irq(codec); + irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); if (irq_val < 0) { dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index 345b3cf..fa32b12 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -16,6 +16,8 @@ #include <sound/soc.h> +int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + #define WM5100_CLK_AIF1 1 #define WM5100_CLK_AIF2 2 #define WM5100_CLK_AIF3 3 -- 1.7.6.3

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[alsa-devel] How to use "hw" or "plughw"?
by Vidblain Amaro 11 Oct '11

11 Oct '11

Hi everbody I have a beginner in this area, i have to develop an audio application to implement filter algorithms to reduce the noisy of an audio I have readed the alsa api, and i decided to partition into 3 threads mi application, one for read samples from an audio file, the second for filter de samples, and the third for playback the audio filtered using snd_pcm_writei, but i am confused about the use "hw" and "plughw" when open de audio device. Some suggestion -- Vidblain "Lo verdadero es demasiado sencillo, pero siempre se llega a ello por lo más complicado."

1 0

[alsa-devel] [PATCH] ASoC: Add missing default for WM5100 Clocking 1
by Mark Brown 11 Oct '11

11 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100-tables.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 960617b..e9ce81a 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -794,6 +794,7 @@ u16 wm5100_reg_defaults[WM5100_MAX_REGISTER + 1] = { [0x0030] = 0x0000, /* R48 - PWM Drive 1 */ [0x0031] = 0x0100, /* R49 - PWM Drive 2 */ [0x0032] = 0x0100, /* R50 - PWM Drive 3 */ + [0x0100] = 0x0002, /* R256 - Clocking 1 */ [0x0101] = 0x0000, /* R257 - Clocking 3 */ [0x0102] = 0x0011, /* R258 - Clocking 4 */ [0x0103] = 0x0011, /* R259 - Clocking 5 */ -- 1.7.6.3

1 0

[alsa-devel] question for the snd_pcm_mmap_writen.
by Anders Gnistrup 11 Oct '11

11 Oct '11

Hi First post to the mail list. I have been reading through the Alsa API and tutorials and it all seems fairly simple. But I still have some questions for the sample format possible to send using the mmap method. First a short description of my problem. I have a embedded device with a 8 channel audio chip. I only have to use 5 of these channels My goal is not to use any cpu cycles on the 3 unused channels I have decided to use the non-interleaved method (application specific, streams comes from different sources in a correct hardware specific format). For this, the function snd_pcm_mmap_writen could be used. When sending data to the device the format should/could be done using a 5 seperate buffers. snd_pcm_sframes_t snd_pcm_mmap_writen (snd_pcm_t * pcm, void ** bufs, snd_pcm_uframes_t size) where *bufs[0] equal to first channel *bufs[1] equals to next etc. Q1) Should/must the **bufs be continuations -> bufs[0][size]==bufs[1][0]??? Q2) Is is possible the set bufs[2] == NULL, to signal that the channel is unused? I think I know the answers. Yes, it should be continues, No data can be set to NULL. Q3) I have tried to find some information about the SND_PCM_ACCESS_MMAP_COMPLEX. What is this??? My best gees is that it is some entirely hardware specific access format? Regards Anders Gnistrup

2 1

[alsa-devel] [PATCH] ASoC: Writing register default value for the reset register
by Axel Lin 11 Oct '11

11 Oct '11

The WM8983 can be reset by performing a write of any value to the software reset register. To avoid writing to the software reset register while resume, we should write the same value in wm8983_reg_defs to software reset register in wm8983_probe(). The write to the reset register is suppressed by the cache restore code when it skips writes of default registers. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8983.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 17f04ec..93ee284 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1007,7 +1007,7 @@ static int wm8983_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0x8983); + ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); return ret; -- 1.7.4.1

3 4

[alsa-devel] [PATCH RESEND] ASoC: Remove impossible case from wm8994_hw_params
by Axel Lin 11 Oct '11

11 Oct '11

We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params. Thus no need to check wm8994-aif3 in wm8994_hw_params. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8994.c | 10 ---------- 1 files changed, 0 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 16542de..68e769e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2235,7 +2235,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; int aif2_reg; @@ -2278,15 +2277,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "AIF2 using split LRCLK\n"); } break; - case 3: - switch (control->type) { - case WM1811: - case WM8958: - aif1_reg = WM8958_AIF3_CONTROL_1; - break; - default: - return 0; - } default: return -EINVAL; } -- 1.7.4.1

2 3

[alsa-devel] [PATCH] ASoC: Fix typo in Kconfig symbol for tlv320aic32x4
by Wolfram Sang 11 Oct '11

11 Oct '11

It is currently named "TVL" instead of "TLV". Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Javier Martin <javier.martin(a)vista-silicon.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 2 +- sound/soc/imx/Kconfig | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d924..d503764 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,7 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER - select SND_SOC_TVL320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -240,7 +240,7 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI -config SND_SOC_TVL320AIC32X4 +config SND_SOC_TLV320AIC32X4 tristate config SND_SOC_TLV320AIC3X diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e..b89a82f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -132,7 +132,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o -obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index bb699bb..62c223f 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -29,7 +29,7 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 - select SND_SOC_TVL320AIC32X4 + select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 -- 1.7.6.3

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Convert to table based init
by Peter Ujfalusi 11 Oct '11

11 Oct '11

Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320dac33.c | 23 +++++++---------------- 1 files changed, 7 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 43ee3b1..3f4920d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -627,18 +627,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"RIGHT_LO", NULL, "Codec Power"}, }; -static int dac33_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets, - ARRAY_SIZE(dac33_dapm_widgets)); - /* set up audio path interconnects */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1451,15 +1439,11 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) } } - snd_soc_add_controls(codec, dac33_snd_controls, - ARRAY_SIZE(dac33_snd_controls)); /* Only add the FIFO controls, if we have valid IRQ number */ if (dac33->irq >= 0) snd_soc_add_controls(codec, dac33_mode_snd_controls, ARRAY_SIZE(dac33_mode_snd_controls)); - dac33_add_widgets(codec); - err_power: return ret; } @@ -1502,6 +1486,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .remove = dac33_soc_remove, .suspend = dac33_soc_suspend, .resume = dac33_soc_resume, + + .controls = dac33_snd_controls, + .num_controls = ARRAY_SIZE(dac33_snd_controls), + .dapm_widgets = dac33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dac33_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; #define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ -- 1.7.7

2 1

[alsa-devel] [PATCH] ASoC: twl4030: Convert to table based init
by Peter Ujfalusi 11 Oct '11

11 Oct '11

Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl4030.c | 21 +++++++-------------- 1 files changed, 7 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 7c244cd..f798247 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1609,17 +1609,6 @@ static const struct snd_soc_dapm_route intercon[] = { }; -static int twl4030_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, twl4030_dapm_widgets, - ARRAY_SIZE(twl4030_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2241,9 +2230,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030_init_chip(codec); - snd_soc_add_controls(codec, twl4030_snd_controls, - ARRAY_SIZE(twl4030_snd_controls)); - twl4030_add_widgets(codec); return 0; } @@ -2269,6 +2255,13 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .reg_cache_size = sizeof(twl4030_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl4030_reg, + + .controls = twl4030_snd_controls, + .num_controls = ARRAY_SIZE(twl4030_snd_controls), + .dapm_widgets = twl4030_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl4030_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int __devinit twl4030_codec_probe(struct platform_device *pdev) -- 1.7.7

2 1

[alsa-devel] [PATCH] ASoC: twl6040: Convert to table based init
by Peter Ujfalusi 11 Oct '11

11 Oct '11

Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 31 ++++++++++--------------------- 1 files changed, 10 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 62edded..93f8a59 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1183,18 +1183,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"AUXR", NULL, "AUXR Playback"}, }; -static int twl6040_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, twl6040_dapm_widgets, - ARRAY_SIZE(twl6040_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - static int twl6040_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1503,16 +1491,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) /* power on device */ ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret) - goto bias_err; - - snd_soc_add_controls(codec, twl6040_snd_controls, - ARRAY_SIZE(twl6040_snd_controls)); - twl6040_add_widgets(codec); - - return 0; + if (!ret) + return 0; -bias_err: + /* Error path */ free_irq(priv->plug_irq, codec); plugirq_err: destroy_workqueue(priv->workqueue); @@ -1544,6 +1526,13 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .reg_cache_size = ARRAY_SIZE(twl6040_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl6040_reg, + + .controls = twl6040_snd_controls, + .num_controls = ARRAY_SIZE(twl6040_snd_controls), + .dapm_widgets = twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int __devinit twl6040_codec_probe(struct platform_device *pdev) -- 1.7.7

2 1

[alsa-devel] LSI HDA Modem is not working on HP new Laptop which has IDT 7605 HDA codec and REDHAT installed on it with ALSA ver 1.0.24.
by Gupta, Suresh 11 Oct '11

11 Oct '11

Modem receive unwanted data on stream 0 even stream is not open by any device. Stream 0 was acquired by IDC 7605 and this codec keep sending something even after closing the stream. On further investigation I found ALSA ver 1.0.24 have new fix "Make converter setups sticky" which changes the semantics of snd_hda_codec_cleanup_stream() not to clean up the stream at that moment but delay the action. This is the reason why Modem getting some unwanted data and stop working. To resolve this issue we need to enable "no_sticky_stream" for these IDT codec. The fix is like putting one line of code "codec->no_sticky_stream =1;" at the end of function "patch_stac92hd83xxx" in file "/sound/pci/hda/patch_sigmatel.c". As I am not much aware of the fix "Make converter setups sticky" and also not know the side effects of above mention fix, please analyze and fix the issue properly. Thanks SuresH

4 15

[alsa-devel] [PATCH] ASoC: Convert Goni to data based DAPM init
by Mark Brown 11 Oct '11

11 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/goni_wm8994.c | 13 +++++-------- 1 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index dcaf656..60ed59b 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -100,14 +100,6 @@ static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - /* add goni specific widgets */ - snd_soc_dapm_new_controls(dapm, goni_dapm_widgets, - ARRAY_SIZE(goni_dapm_widgets)); - - /* set up goni specific audio routes */ - snd_soc_dapm_add_routes(dapm, goni_dapm_routes, - ARRAY_SIZE(goni_dapm_routes)); - /* set endpoints to not connected */ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); @@ -256,6 +248,11 @@ static struct snd_soc_card goni = { .name = "goni", .dai_link = goni_dai, .num_links = ARRAY_SIZE(goni_dai), + + .dapm_widgets = goni_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets), + .dapm_routes = goni_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(goni_dapm_routes), }; static int __init goni_init(void) -- 1.7.6.3

2 1

Re: [alsa-devel] [PATCH v2 3/6] MFD: twl6040: function to query the vibra status for clients
by Ujfalusi, Peter 11 Oct '11

11 Oct '11

Hi Samuel, On Mon, Oct 10, 2011 at 10:05 PM, Samuel Ortiz <samuel.ortiz(a)intel.com> wrote: > Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> Thank you. > I assume those 2 go upstream through the ASoC tree, right ? Yes, that's the plan. -- Péter

1 0

Re: [alsa-devel] Using spdiff for backporting
by Luis R. Rodriguez 10 Oct '11

10 Oct '11

On Tue, Oct 4, 2011 at 2:37 AM, Jesper Andersen <jespera(a)gmail.com> wrote: > Hi > > Somehow, I think the email below was never sent? Could either of you confirm my suspicion? I just checked and I did get the original e-mail. Apologies for the late reply. Below my reply. > ====== Original email ====== > > Here are a two results that I have found using spdiff on some of the > patches from compat-wireless. > > 01-netdev: > @@ > struct net_device * X5; > const struct net_device_ops X6; > @@ > - X5->netdev_ops=&X6; > + netdev_attach_ops(X5,&X6); > > 27-hermes-read-pda: > @@ > struct hermes * X0; > @@ > - X0->ops->read_pda > + X0->ops->read_pda_h > > Basically, I have done the following > > - clone the linux-next tree > - apply a patch from compat-wireless/patches > - git commit > - extract pairs of original and modified files affected by the commit > (I call this set of pairs a "change set") > - apply spdiff to the extracted pairs > - adjust parameters to spdiff until something reasonable comes out > > For spdiff to work in its current form we need a change set with at > least two similar modifications. For the netdev-ops case we actually > have many changes matching "X5->netdev_ops=&X6" and seemingly those > locations are all rewritten into: "netdev_attach_ops(X5,&X6)". > > For many of the patches in compat-wireless/patches, however such a > scenario is not present. For example 03-rfkill consists of just one > change of the string "rfkill" to "rfkill_backport". In this case spdiff > cannot really infer anything useful. Ah I see. > The question is: given that spdiff in its current form really needs at > least two example changes, could you perhaps point me at which of the > patches in compat-wireless/patches should contain such things? Sure thing. I'll review that and get back to you. > Also, in the slides from the plumbers-conf. I think I saw it mentioned > that there's also an alsa-compat git-tree somewhere. Maybe that is a > more compelling usecase? I was unable to find it though so I'd > appreciate a link. It was on kernel.org, but if it got rm -rf'd then its gone as I do not have a local copy. The compat-alsa stuff though was determined to be superflous with the ALSA's team's own backport work which is currently independent. My hope though is to unify these through the compat.git / compat-kernel (currently just called compat-wireless) effort. > A way that we maybe could get some more similar changes would be to > focus on finding a patch for a specific version. E.g. for > 09-threaded-irq there's a lot if #ifdefs for versions less than 2.6.31. > In that case it seems for the lower version numbers, some code is added > and for higher versions some other code has to be present. Instead, we > could try to find a patch from the changes between >2.6.31 and <2.6.31 > perhaps? Not sure I get this, could you elaborate a bit more? Are you saying to cheat and double up the ifdef's for at least two older kernels or newer kernels so that spatch gets two "change sets"? > Or maybe you (Luis) have a better suggestion for how spdiff could be > used. I am very open to modifications required specifically for doing > backports work, I just need to get an idea for what would be useful to > be able to infer. :-) I'm perfectly happy with living with spdiff only helping with large "change sets", and dealing with the rest of the patches manually. Likely spdiff can help when we decide we want to add backport support for a new subsystem, so we'd just apply the spatch's generated to the new directories we want to consider, and then just require manual work for what is left. Luis

2 2

[alsa-devel] [PATCH v2 0/9] ASoC: OMAP machine driver cleanups
by Peter Ujfalusi 10 Oct '11

10 Oct '11

Hello, Some cleanups for the OMAP ASoC machine drivers: - do not call soc_dapm_sync from dai_link->init callback since it is useless - Convert drivers to use table based DAPM initialization - Remove dapm_enable_pin calls from dai_link->init callback since pins are enabled by default Changelog since v1: - Do not touch ams-delta just remove the dapm_sync call. - Do not convert soc_dapm_sync calls to dapm_new_widgets call Regards, Peter --- Peter Ujfalusi (9): ASoC: OMAP machines: Remove soc_dapm_sync() call from init ASoC: am3517evm: Let core to deal with the DAPM widgets ASoC: n810: Let the core to register DAPM widgets/routes and controls ASoC: osk5912: Let core to deal with the DAPM widgets ASoC: sdp3430: Let core to deal with the DAPM widgets ASoC: zoom2: Let core to deal with the DAPM widgets ASoC: zoom2: No need to call dapm_pin_enable at init time ASoC: sdp4430: Let core to deal with the DAPM widgets ASoC: sdp4430: No need to call dapm_pin_enable at init time sound/soc/omap/am3517evm.c | 28 +++++----------------------- sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/n810.c | 23 +++++++---------------- sound/soc/omap/omap3pandora.c | 8 ++------ sound/soc/omap/osk5912.c | 27 +++++---------------------- sound/soc/omap/rx51.c | 2 -- sound/soc/omap/sdp3430.c | 18 +++++------------- sound/soc/omap/sdp4430.c | 27 +++++---------------------- sound/soc/omap/zoom2.c | 26 ++++++-------------------- 9 files changed, 35 insertions(+), 125 deletions(-) -- 1.7.7

2 10

[alsa-devel] ASoC updates for 3.2
by Mark Brown 10 Oct '11

10 Oct '11

The following changes since commit 734787550a60b768b675c26f93d134f6dc370bb5: ASoC: wm8995: Slightly optimize configure_clock (2011-10-04 13:09:55 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (24): ASoC: wm8994: Fix setting rate_reg for wm8994-aif2 ASoC: Avoid writing to WM8971_RESET in wm8971_resume ASoC: wm8750: Use snd_soc_update_bits for read-modify-write ASoC: wm8988: Use snd_soc_update_bits for read-modify-write ASoC: wm8711: Use snd_soc_update_bits for read-modify-write ASoC: Remove unneeded hw_write initialisation in ak4671 ASoC: Remove unneeded hw_write initialisation in wm8523 ASoC: Remove unused function declaration in imx-ssi.h ASoC: imx: Remove unused variable 'dai' ASoC: Remove unused variable 'wm9090' in wm9090_probe ASoC: wm8990: Convert to snd_soc_cache_sync for sync reg_cache with the hardware ASoC: wm8990: Remove incorrect comments ASoC: Return early with -EINVAL if invalid dai format is detected ASoC: wm8510: Convert to snd_soc_cache_sync ASoC: wm8711: Convert to snd_soc_cache_sync ASoC: wm8731: Convert to snd_soc_cache_sync ASoC: wm8750: Convert to snd_soc_cache_sync ASoC: wm8776: Convert to snd_soc_cache_sync ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_cache_sync ASoC: wm8971: Convert to snd_soc_cache_sync ASoC: wm8974: Convert to snd_soc_cache_sync ASoC: wm8988: Convert to snd_soc_cache_sync ASoC: ak4535: convert to soc-cache Mark Brown (32): Merge branch 'for-3.1' into for-3.2 ASoC: Remove direct register cache accesses from WM8962 driver regulator: Fix return code from regulator_disable_deferred() ASoC: Factor write of widget power out into a separate function ASoC: Move bias level decision into main dapm_power_widgets() ASoC: Ensure all DAPM widgets have a power check callback ASoC: Factor out widget power check operation ASoC: Mark headphone, mic, speaker and line widgets as always connected ASoC: Do DAPM power checks only for widgets changed since last run ASoC: Reduce the number of neigbours we mark dirty when updating power ASoC: Add verbose debugging showing why widgets get marked dirty ASoC: Only run power_check() on a widget once per run ASoC: Don't mark the outputs of supplies as dirty on state changes ASoC: Stop checking for supplied widgets after we find the first ASoC: Use dapm_mark_dirty() for new DAPM widgets for consistency ASoC: Remove references to linux(a)wolfsonmicro.com ASoC: Ensure DAPM widgets are set up before we sync jacks Merge branch 'for-3.1' into for-3.2 regulator: Add WM1811 support ASoC: Suppress early calls to snd_soc_dapm_sync() ASoC: Remove redundant snd_soc_dapm_sync() calls from machine drivers ASoC: Ensure all DAPM widgets are instantiated with the card ASoC: Assign power_check when we allocate DAPM widgets ASoC: Cache connected input and output recursions ASoC: Squash error codes from regmap down to -1 on read ASoC: Fix DAPM sync for TLV320AIC3x custom DAPM widget ASoC: Convert Simtec machines to table based DAPM init ASoC: Convert H1940 to table based init ASoC: Convert RX1950 to table based init ASoC: Convert SmartQ to table based init ASoC: Convert SMDK WM8580 to table based DAPM init ASoC: Convert Jive to table based init Michael Opdenacker (1): ASoC: fix checkpatch.pl error in omap-mcbsp Peter Ujfalusi (15): ASoC: core: Introduce SOC_DOUBLE_VALUE macro ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro ASoC: twl6040: Simplify custom put_volsw callback ASoC: Consolidate use of controls with custom get/put function ASoC: twl6040: Simplify custom get_volsw callback ASoC: twl6040: Prepare for core put_volsw/volsw_2r merger ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions ASoC: core: Combine snd_soc_get_volsw/get_volsw_2r functions ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions ASoC: twl6040: Simply call snd_soc_put_volsw form the custom code ASoC: core: Remove snd_soc_put_volsw_2r definition ASoC: twl6040: Warn user in twl6040_put_volsw for error case ASoC: Replace remaining use of *_volsw_2r with *_volsw ASoC: fix codec breakage caused by the volsw/volsw_2r merger Wolfram Sang (1): ASoC: tlv320aic32x4 fix initialization of micpga routing drivers/regulator/core.c | 9 +- drivers/regulator/wm8994-regulator.c | 13 + include/sound/soc-dapm.h | 8 + include/sound/soc.h | 85 ++++--- sound/soc/atmel/playpaq_wm8510.c | 1 - sound/soc/atmel/sam9g20_wm8731.c | 2 - sound/soc/atmel/snd-soc-afeb9260.c | 2 - sound/soc/codecs/ak4535.c | 100 ++------ sound/soc/codecs/ak4671.c | 2 - sound/soc/codecs/cs4270.c | 8 +- sound/soc/codecs/cs42l51.c | 9 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/tlv320aic23.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/codecs/tlv320aic3x.c | 4 + sound/soc/codecs/twl4030.c | 48 +--- sound/soc/codecs/twl6040.c | 76 ++---- sound/soc/codecs/wm8350.c | 43 ++-- sound/soc/codecs/wm8510.c | 13 +- sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 38 +-- sound/soc/codecs/wm8711.c | 19 +- sound/soc/codecs/wm8731.c | 15 +- sound/soc/codecs/wm8750.c | 42 +--- sound/soc/codecs/wm8776.c | 16 +- sound/soc/codecs/wm8940.c | 46 ++-- sound/soc/codecs/wm8960.c | 15 +- sound/soc/codecs/wm8962.c | 21 +- sound/soc/codecs/wm8971.c | 15 +- sound/soc/codecs/wm8974.c | 15 +- sound/soc/codecs/wm8988.c | 18 +-- sound/soc/codecs/wm8990.c | 36 ++-- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm9090.c | 1 - sound/soc/codecs/wm_hubs.c | 20 +- sound/soc/davinci/davinci-evm.c | 2 - sound/soc/imx/imx-pcm-fiq.c | 1 - sound/soc/imx/imx-ssi.c | 1 - sound/soc/imx/imx-ssi.h | 6 - sound/soc/kirkwood/kirkwood-t5325.c | 2 - sound/soc/mid-x86/mfld_machine.c | 2 - sound/soc/omap/omap-mcbsp.c | 3 +- sound/soc/pxa/corgi.c | 1 - sound/soc/pxa/e740_wm9705.c | 2 - sound/soc/pxa/e750_wm9705.c | 2 - sound/soc/pxa/e800_wm9712.c | 1 - sound/soc/pxa/magician.c | 1 - sound/soc/pxa/mioa701_wm9713.c | 1 - sound/soc/pxa/palm27x.c | 4 - sound/soc/pxa/saarb.c | 4 - sound/soc/pxa/tavorevb3.c | 4 - sound/soc/pxa/tosa.c | 1 - sound/soc/pxa/z2.c | 4 - sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/goni_wm8994.c | 2 - sound/soc/samsung/h1940_uda1380.c | 19 +- sound/soc/samsung/jive_wm8750.c | 17 +- sound/soc/samsung/neo1973_wm8753.c | 4 - sound/soc/samsung/rx1950_uda1380.c | 33 +-- sound/soc/samsung/s3c24xx_simtec_hermes.c | 11 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 11 +- sound/soc/samsung/smartq_wm8987.c | 25 +-- sound/soc/samsung/smdk_wm8580.c | 45 +--- sound/soc/samsung/smdk_wm8994.c | 2 - sound/soc/sh/sh7760-ac97.c | 7 - sound/soc/soc-core.c | 166 +++--------- sound/soc/soc-dapm.c | 346 +++++++++++++++++------- sound/soc/soc-io.c | 2 +- sound/soc/soc-jack.c | 2 + sound/soc/tegra/tegra_wm8903.c | 2 - sound/soc/tegra/trimslice.c | 2 - 72 files changed, 601 insertions(+), 903 deletions(-)

2 2

[alsa-devel] [PATCH] ASoC: Instantiate card widgets immediately
by Mark Brown 10 Oct '11

10 Oct '11

This ensures they are available prior to the card late_probe(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8dc9aba..d6af52b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1430,6 +1430,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + snd_soc_dapm_new_widgets(&card->dapm); + for (i = 0; i < card->num_links; i++) { dai_link = &card->dai_link[i]; -- 1.7.6.3

1 0

[alsa-devel] amixer cget :Assertion `ctl && info && (info->id.name[0] || info->id.numid)' failed. Aborted
by Raymond Yau 10 Oct '11

10 Oct '11

There is assertion when specify iface=PCM or iface=PCM,name='' when using amixer -c0 cget according to the synthax [[iface=<iface>,][name='name',][index=<index>,][device=<device>,][subdevice=<subdevice>]]|[numid=<numid>] amixer -c0 cget iface=PCM amixer: control.c:258: snd_ctl_elem_info: Assertion `ctl && info && (info->id.name[0] || info->id.numid)' failed. Aborted amixer -c0 cget iface=PCM,name='' amixer: control.c:258: snd_ctl_elem_info: Assertion `ctl && info && (info->id.name[0] || info->id.numid)' failed. Aborted amixer -c0 cget Specify a full control identifier: [[iface=<iface>,][name='name',][index=<index>,][device=<device>,][subdevice=<subdevice>]]|[numid=<numid>] amixer -c0 cget iface=PCM,name='ILD' amixer: Cannot find the given element from control hw:0 amixer -c0 cget iface=PCM,name='ILD',device=2 numid=45,iface=PCM,name='ILD',device=2 ; type=INTEGER,access=rw------,values=2,min=0,max=2147483647,step=0 : values=0,0

3 6

[alsa-devel] [PATCH] ALSA: remove the dead code from snd_pcm_open_file()
by Feng Tang 10 Oct '11

10 Oct '11

The rpcm_file parameter is never used in current ALSA code, so remove it to make it cleaner. Signed-off-by: Feng Tang <feng.tang(a)intel.com> --- sound/core/pcm_native.c | 12 +++--------- 1 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1c6be91..2d3af5d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2058,16 +2058,12 @@ EXPORT_SYMBOL(snd_pcm_open_substream); static int snd_pcm_open_file(struct file *file, struct snd_pcm *pcm, - int stream, - struct snd_pcm_file **rpcm_file) + int stream) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; int err; - if (rpcm_file) - *rpcm_file = NULL; - err = snd_pcm_open_substream(pcm, stream, file, &substream); if (err < 0) return err; @@ -2083,8 +2079,7 @@ static int snd_pcm_open_file(struct file *file, substream->pcm_release = pcm_release_private; } file->private_data = pcm_file; - if (rpcm_file) - *rpcm_file = pcm_file; + return 0; } @@ -2113,7 +2108,6 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) { int err; - struct snd_pcm_file *pcm_file; wait_queue_t wait; if (pcm == NULL) { @@ -2131,7 +2125,7 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) add_wait_queue(&pcm->open_wait, &wait); mutex_lock(&pcm->open_mutex); while (1) { - err = snd_pcm_open_file(file, pcm, stream, &pcm_file); + err = snd_pcm_open_file(file, pcm, stream); if (err >= 0) break; if (err == -EAGAIN) { -- 1.7.1

2 1

[alsa-devel] [PATCH 1/5] ASoC: Convert H1940 to table based init
by Mark Brown 10 Oct '11

10 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/h1940_uda1380.c | 17 +++++------------ 1 files changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index cdaed90..03cfa5f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -183,18 +183,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Add h1940 specific widgets */ - err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - if (err) - return err; - - /* Set up h1940 specific audio path audio_mapnects */ - err = snd_soc_dapm_add_routes(dapm, audio_map, - ARRAY_SIZE(audio_map)); - if (err) - return err; - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); @@ -229,6 +217,11 @@ static struct snd_soc_card h1940_asoc = { .name = "h1940", .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), + + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int __init h1940_init(void) -- 1.7.6.3

2 11

[alsa-devel] [PATCH v3 00/10] Convert wm8xxx codec drivers to use snd_soc_write/snd_soc_cache_sync
by Axel Lin 10 Oct '11

10 Oct '11

This serial converts the wm8xxx codec drivers to use snd_soc_cache_sync. I don't have these hardware, but I assume Mark has all these hardware. These patches are UNTESTED. I'd appreciate if someone can test it. (Current code is broken anyway, thus we need convert to use new API. ) v3: use snd_soc_cache_sync to replace customized xxxx_sync_cache() calls. Drop wm8990 patch in v2, which is already merged. I also change the subject line back to be the same as v1. ( we are using snd_soc_cache_sync now, not snd_soc_write ) v2: For 7, 9 CODECs that are write only, convert to using cache_bypass + snd_soc_write v2 also includes wm8990 that I had sent separately. Axel Lin (10): ASoC: wm8510: Convert to snd_soc_cache_sync ASoC: wm8711: Convert to snd_soc_cache_sync ASoC: wm8731: Convert to snd_soc_cache_sync ASoC: wm8750: Convert to snd_soc_cache_sync ASoC: wm8776: Convert to snd_soc_cache_sync ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_cache_sync ASoC: wm8971: Convert to snd_soc_cache_sync ASoC: wm8974: Convert to snd_soc_cache_sync ASoC: wm8988: Convert to snd_soc_cache_sync sound/soc/codecs/wm8510.c | 13 +---------- sound/soc/codecs/wm8711.c | 15 ++----------- sound/soc/codecs/wm8731.c | 15 +------------ sound/soc/codecs/wm8750.c | 16 +------------- sound/soc/codecs/wm8776.c | 16 +------------- sound/soc/codecs/wm8940.c | 46 +++++++++++++++++++++----------------------- sound/soc/codecs/wm8960.c | 14 +++--------- sound/soc/codecs/wm8971.c | 15 ++----------- sound/soc/codecs/wm8974.c | 13 +---------- sound/soc/codecs/wm8988.c | 17 +-------------- 10 files changed, 44 insertions(+), 136 deletions(-) -- 1.7.4.1

2 11

[alsa-devel] [RFC][PATCH] ASoC: ak4535: convert to soc-cache
by Axel Lin 10 Oct '11

10 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, this patch is UNTESTED!! ( But current code is broken anyway ) I'd appreciate if someone can test it. Axel sound/soc/codecs/ak4535.c | 100 ++++++++------------------------------------- 1 files changed, 18 insertions(+), 82 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1a214e..68df32d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -34,7 +34,6 @@ struct ak4535_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -47,63 +46,6 @@ static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { 0x0000, 0x0000, 0x0057, 0x0000, }; -/* - * read ak4535 register cache - */ -static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4535 register cache - */ -static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the AK4535 register space - */ -static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4535 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - ak4535_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static int ak4535_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4535_CACHEREGNUM; i++) - r |= ak4535_write(codec, i, cache[i]); - - return r; -}; - static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; @@ -304,7 +246,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; if (rate) @@ -323,7 +265,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, } /* set rate */ - ak4535_write(codec, AK4535_MODE2, mode2); + snd_soc_write(codec, AK4535_MODE2, mode2); return 0; } @@ -348,44 +290,37 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, /* use 32 fs for BCLK to save power */ mode1 |= 0x4; - ak4535_write(codec, AK4535_MODE1, mode1); + snd_soc_write(codec, AK4535_MODE1, mode1); return 0; } static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); + u16 mute_reg = snd_soc_read(codec, AK4535_DAC); if (!mute) - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg & ~0x20); else - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg | 0x20); return 0; } static int ak4535_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 i, mute_reg; - switch (level) { case SND_SOC_BIAS_ON: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0); break; case SND_SOC_BIAS_PREPARE: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0x20); break; case SND_SOC_BIAS_STANDBY: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i | 0x80); - i = ak4535_read_reg_cache(codec, AK4535_PM2); - ak4535_write(codec, AK4535_PM2, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0x80); + snd_soc_update_bits(codec, AK4535_PM2, 0x80, 0); break; case SND_SOC_BIAS_OFF: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0); break; } codec->dapm.bias_level = level; @@ -428,7 +363,7 @@ static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state) static int ak4535_resume(struct snd_soc_codec *codec) { - ak4535_sync(codec); + snd_soc_cache_sync(codec); ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -436,11 +371,15 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); + int ret; printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); - codec->control_data = ak4535->control_data; - + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -461,8 +400,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .remove = ak4535_remove, .suspend = ak4535_suspend, .resume = ak4535_resume, - .read = ak4535_read_reg_cache, - .write = ak4535_write, .set_bias_level = ak4535_set_bias_level, .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), @@ -485,7 +422,6 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4535); - ak4535->control_data = i2c; ak4535->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

2 1

[alsa-devel] [PATCH 0/2] omap: sound: style and warning fixes
by Michael Opdenacker 10 Oct '11

10 Oct '11

These patches apply to Tony Lindgren's linux-omap tree (they can also be applied to Linus' tree) They fix a warning at compile time and a coding style error.

3 5

[alsa-devel] [PATCH] ASoC: Fix DAPM sync for TLV320AIC3x custom DAPM widget
by Mark Brown 10 Oct '11

10 Oct '11

We really should be doing this in the core, not in a driver... Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 +++ sound/soc/codecs/tlv320aic3x.c | 4 ++++ sound/soc/soc-dapm.c | 3 ++- 3 files changed, 9 insertions(+), 1 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e2853da..5b2202f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -381,6 +381,9 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* Mostly internal - should not normally be used */ +void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d877b39..be55b7f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -197,6 +197,10 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, else /* old connection must be powered down */ path->connect = invert ? 1 : 0; + + dapm_mark_dirty(path->source, "tlv320aic3x source"); + dapm_mark_dirty(path->sink, "tlv320aic3x sink"); + break; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e49c56d..b9836ab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,7 +124,7 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", @@ -132,6 +132,7 @@ static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } +EXPORT_SYMBOL_GPL(dapm_mark_dirty); /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( -- 1.7.6.3

2 1

[alsa-devel] [PATCH] ASoC: ams-delta: Let the core to register DAPM widgets/routes and controls
by Peter Ujfalusi 10 Oct '11

10 Oct '11

Pass the DAPM widgets/routes and static controls via the snd_soc_card struct to core. In this way the machine driver does not need to handle the DAPM widgets/routes. Move the setup for initial pin constellation to late_probe time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/ams-delta.c | 56 +++++++++++++++---------------------------- 1 files changed, 20 insertions(+), 36 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 10d3d41..2153a1c 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -536,48 +536,23 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Register optional line discipline for over the modem control */ ret = tty_register_ldisc(N_V253, &cx81801_ops); - if (ret) { + if (ret) dev_warn(card->dev, "Failed to register line discipline, " "will continue without any controls.\n"); - return 0; - } - - /* Add board specific DAPM widgets and routes */ - ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets, - ARRAY_SIZE(ams_delta_dapm_widgets)); - if (ret) { - dev_warn(card->dev, - "Failed to register DAPM controls, " - "will continue without any.\n"); - return 0; - } - ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map, - ARRAY_SIZE(ams_delta_audio_map)); - if (ret) { - dev_warn(card->dev, - "Failed to set up DAPM routes, " - "will continue with codec default map.\n"); - return 0; - } + return 0; +} +static int ams_delta_late_probe(struct snd_soc_card *card) +{ /* Set up initial pin constellation */ - snd_soc_dapm_disable_pin(dapm, "Mouthpiece"); - snd_soc_dapm_enable_pin(dapm, "Earpiece"); - snd_soc_dapm_enable_pin(dapm, "Microphone"); - snd_soc_dapm_disable_pin(dapm, "Speaker"); - snd_soc_dapm_disable_pin(dapm, "AGCIN"); - snd_soc_dapm_disable_pin(dapm, "AGCOUT"); - snd_soc_dapm_new_widgets(dapm); - - /* Add virtual switch */ - ret = snd_soc_add_controls(codec, ams_delta_audio_controls, - ARRAY_SIZE(ams_delta_audio_controls)); - if (ret) - dev_warn(card->dev, - "Failed to register audio mode control, " - "will continue without it.\n"); + snd_soc_dapm_disable_pin(&card->dapm, "Mouthpiece"); + snd_soc_dapm_enable_pin(&card->dapm, "Earpiece"); + snd_soc_dapm_enable_pin(&card->dapm, "Microphone"); + snd_soc_dapm_disable_pin(&card->dapm, "Speaker"); + snd_soc_dapm_disable_pin(&card->dapm, "AGCIN"); + snd_soc_dapm_disable_pin(&card->dapm, "AGCOUT"); return 0; } @@ -600,6 +575,15 @@ static struct snd_soc_card ams_delta_audio_card = { .dai_link = &ams_delta_dai_link, .num_links = 1, .set_bias_level = ams_delta_set_bias_level, + + .controls = ams_delta_audio_controls, + .num_controls = ARRAY_SIZE(ams_delta_audio_controls), + .dapm_widgets = ams_delta_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ams_delta_dapm_widgets), + .dapm_routes = ams_delta_audio_map, + .num_dapm_routes = ARRAY_SIZE(ams_delta_audio_map), + + .late_probe = ams_delta_late_probe, }; /* Module init/exit */ -- 1.7.7

2 2

[alsa-devel] [PATCH] ASoC: Squash error codes from regmap down to -1 on read
by Mark Brown 09 Oct '11

09 Oct '11

The ASoC code always uses -1 as the error code due to reporting errors in band with the value. Ensure we don't confuse anything by making sure we don't pass actual error codes back into the rest of the code on read. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-io.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index d6535b8..c8610cb 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -56,7 +56,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) if (ret == 0) return val; else - return ret; + return -1; } ret = snd_soc_cache_read(codec, reg, &val); -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC: Cache connected input and output recursions
by Mark Brown 09 Oct '11

09 Oct '11

The number of connected input and output endpoints for a given widgets can't change during a DAPM run so there is no need to redo the recursion through branches of the tree we've already visited. Doing this on one of my test systems gives an improvement of: Power Path Neighbour Before: 63 607 731 After: 63 141 181 which scales up well as more widgets are involved in paths. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 + sound/soc/soc-dapm.c | 60 ++++++++++++++++++++++++++++++++++----------- 2 files changed, 47 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5b2202f..17a4c17 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -498,6 +498,8 @@ struct snd_soc_dapm_widget { /* used during DAPM updates */ struct list_head power_list; struct list_head dirty; + int inputs; + int outputs; }; struct snd_soc_dapm_update { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 24e5d45..f42e8b9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -666,6 +666,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->outputs >= 0) + return widget->outputs; + DAPM_UPDATE_STAT(widget, path_checks); if (widget->id == snd_soc_dapm_supply) @@ -674,21 +677,29 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - if (widget->active) - return snd_soc_dapm_suspend_check(widget); + if (widget->active) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } default: break; } if (widget->connected) { /* connected pin ? */ - if (widget->id == snd_soc_dapm_output && !widget->ext) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_output && !widget->ext) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } /* connected jack or spk ? */ - if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || - (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_hp || + widget->id == snd_soc_dapm_spk || + (widget->id == snd_soc_dapm_line && + !list_empty(&widget->sources))) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; + } } list_for_each_entry(path, &widget->sinks, list_source) { @@ -706,6 +717,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } } + widget->outputs = con; + return con; } @@ -718,6 +731,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->inputs >= 0) + return widget->inputs; + DAPM_UPDATE_STAT(widget, path_checks); if (widget->id == snd_soc_dapm_supply) @@ -727,25 +743,35 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - if (widget->active) - return snd_soc_dapm_suspend_check(widget); + if (widget->active) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } default: break; } if (widget->connected) { /* connected pin ? */ - if (widget->id == snd_soc_dapm_input && !widget->ext) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_input && !widget->ext) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } /* connected VMID/Bias for lower pops */ - if (widget->id == snd_soc_dapm_vmid) - return snd_soc_dapm_suspend_check(widget); + if (widget->id == snd_soc_dapm_vmid) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } /* connected jack ? */ if (widget->id == snd_soc_dapm_mic || - (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks))) - return snd_soc_dapm_suspend_check(widget); + (widget->id == snd_soc_dapm_line && + !list_empty(&widget->sinks))) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } + } list_for_each_entry(path, &widget->sources, list_sink) { @@ -763,6 +789,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } } + widget->inputs = con; + return con; } @@ -1336,6 +1364,8 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(w, &card->widgets, list) { w->power_checked = false; + w->inputs = -1; + w->outputs = -1; } /* Check which widgets we need to power and store them in -- 1.7.6.3

1 0

[alsa-devel] ALSA: hda - Fix a regression of the position-buffer check
by David Henningsson 09 Oct '11

09 Oct '11

Hi Takashi, The offending commit a810364a042 seems to be in v3.0-rc2 (according to git describe), so is there a reason why the below is not sent to 3.0 stable tree? commit 798cb7e897210dbf827f69a302c0967e0380ebac Author: Takashi Iwai <tiwai(a)suse.de> Date: Fri Sep 30 08:52:26 2011 +0200 ALSA: hda - Fix a regression of the position-buffer check The commit a810364a0424c297242c6c66071a42f7675a5568 ALSA: hda - Handle -1 as invalid position, too caused a regression on some machines that require the position-buffer instead of LPIB, e.g. resulting in noises with mic recording with PulseAudio. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 2

[alsa-devel] [PATCH 1/2] ASoC: Ensure all DAPM widgets are instantiated with the card
by Mark Brown 08 Oct '11

08 Oct '11

Specifically for the widgets added by machine driver late probe functions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b65e3d4..8dc9aba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1478,6 +1478,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + snd_soc_dapm_new_widgets(&card->dapm); + ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); -- 1.7.6.3

1 1

[alsa-devel] [PATCH] ASoC: Remove redundant snd_soc_dapm_sync() calls from machine drivers
by Mark Brown 08 Oct '11

08 Oct '11

The core will sync DAPM as part of the card initialization, there is no need for machine drivers to do so during their setup. OMAP drivers are omitted as I know Peter already has patches for them. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/atmel/playpaq_wm8510.c | 1 - sound/soc/atmel/sam9g20_wm8731.c | 2 -- sound/soc/atmel/snd-soc-afeb9260.c | 2 -- sound/soc/davinci/davinci-evm.c | 2 -- sound/soc/kirkwood/kirkwood-t5325.c | 2 -- sound/soc/mid-x86/mfld_machine.c | 2 -- sound/soc/pxa/corgi.c | 1 - sound/soc/pxa/e740_wm9705.c | 2 -- sound/soc/pxa/e750_wm9705.c | 2 -- sound/soc/pxa/e800_wm9712.c | 1 - sound/soc/pxa/magician.c | 1 - sound/soc/pxa/mioa701_wm9713.c | 1 - sound/soc/pxa/palm27x.c | 4 ---- sound/soc/pxa/saarb.c | 4 ---- sound/soc/pxa/tavorevb3.c | 4 ---- sound/soc/pxa/tosa.c | 1 - sound/soc/pxa/z2.c | 4 ---- sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/goni_wm8994.c | 2 -- sound/soc/samsung/h1940_uda1380.c | 2 -- sound/soc/samsung/jive_wm8750.c | 1 - sound/soc/samsung/neo1973_wm8753.c | 4 ---- sound/soc/samsung/rx1950_uda1380.c | 2 -- sound/soc/samsung/s3c24xx_simtec_hermes.c | 1 - sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 1 - sound/soc/samsung/smartq_wm8987.c | 4 ---- sound/soc/samsung/smdk_wm8580.c | 6 ------ sound/soc/samsung/smdk_wm8994.c | 2 -- sound/soc/sh/sh7760-ac97.c | 7 ------- sound/soc/tegra/tegra_wm8903.c | 2 -- sound/soc/tegra/trimslice.c | 2 -- 31 files changed, 0 insertions(+), 73 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 2909bfa..73ae99a 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -338,7 +338,6 @@ static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd) /* always connected pins */ snd_soc_dapm_enable_pin(dapm, "Int Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_sync(dapm); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index bad3aa1..0377c54 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -173,8 +173,6 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* always connected */ snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index 5e4d499..d427e92 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -117,8 +117,6 @@ static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fe79842..f78c3f0 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -150,8 +150,6 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index c8d2195..c772b3c 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -79,8 +79,6 @@ static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 277ca626..cca693a 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -234,7 +234,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* always connected */ snd_soc_dapm_enable_pin(dapm, "Headphones"); snd_soc_dapm_enable_pin(dapm, "Mic"); - snd_soc_dapm_sync(dapm); ret_val = snd_soc_add_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); @@ -252,7 +251,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - snd_soc_dapm_sync(dapm); /* Headset and button jack detection */ ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 28757fb..b0e2fb7 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -299,7 +299,6 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* Set up corgi specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index dc65650..35ed7eb8c 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -108,8 +108,6 @@ static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 51897fc..ce5f056 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -90,8 +90,6 @@ static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 053ed20..6a8f38b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -80,7 +80,6 @@ static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(e800_dapm_widgets)); snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 098e5f4..e79f516 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -423,7 +423,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) /* Set up magician specific audio path interconnects */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 38ca675..0b8d1ee 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -151,7 +151,6 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Front Mic"); snd_soc_dapm_enable_pin(dapm, "GSM Line In"); snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 504e400..7edc1fb 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -107,10 +107,6 @@ static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "PHONE"); snd_soc_dapm_nc_pin(dapm, "MIC2"); - err = snd_soc_dapm_sync(dapm); - if (err) - return err; - /* Jack detection API stuff */ err = snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hs_jack); diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 9595189..d9467a2 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -146,10 +146,6 @@ static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index f881f65..eeec892 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -146,10 +146,6 @@ static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); - if (ret) - return ret; - /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 9a23513..620fc69 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -211,7 +211,6 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) /* set up tosa specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 4b81ffd..b311ffe 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -161,10 +161,6 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) /* Set up z2 specific audio paths */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - ret = snd_soc_dapm_sync(dapm); - if (ret) - goto err; - /* Jack detection API stuff */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 2b8350b..580aae3 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -87,7 +87,6 @@ static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headphone"); snd_soc_dapm_enable_pin(dapm, "Headset Earpiece"); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index dcaf656..7711650 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -121,8 +121,6 @@ static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "SPKOUTRP"); } - snd_soc_dapm_sync(dapm); - /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 374ac2e..cdaed90 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -199,8 +199,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack); diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index b4e3d01..5a1e0cb 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -121,7 +121,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) } snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 16152ed..7207189 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -367,8 +367,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_soc_dapm_sync(dapm); - return 0; } @@ -409,8 +407,6 @@ static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); snd_soc_dapm_ignore_suspend(dapm, "Headphone"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index de1c509..d197100 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -239,8 +239,6 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack); diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 2422117..5a9d495 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -77,7 +77,6 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 57aee5e..203bd7a 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -66,7 +66,6 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index ebac492..16bf48f 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -179,10 +179,6 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Internal Mic"); snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - err = snd_soc_dapm_sync(dapm); - if (err) - return err; - /* Headphone jack detection */ err = snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &smartq_jack); diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index dabd185..9de6b13 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -174,9 +174,6 @@ static int smdk_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) */ snd_soc_dapm_disable_pin(dapm, "MicIn"); - /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); - return 0; } @@ -192,9 +189,6 @@ static int smdk_wm8580_init_paifrx(struct snd_soc_pcm_runtime *rtd) /* Set up PAIFRX audio path */ snd_soc_dapm_add_routes(dapm, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 45fbe2b..f75e439 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -117,8 +117,6 @@ static int smdk_wm8994_init_paiftx(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "IN1RP"); snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 917d3ce..c62ae68 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,12 +20,6 @@ extern struct snd_soc_dai_driver sh4_hac_dai[2]; extern struct snd_soc_platform_driver sh7760_soc_platform; -static int machine_init(struct snd_soc_pcm_runtime *rtd) -{ - snd_soc_dapm_sync(&rtd->codec->dapm); - return 0; -} - static struct snd_soc_dai_link sh7760_ac97_dai = { .name = "AC97", .stream_name = "AC97 HiFi", @@ -33,7 +27,6 @@ static struct snd_soc_dai_link sh7760_ac97_dai = { .codec_dai_name = "ac97-hifi", .platform_name = "sh7760-pcm-audio", .codec_name = "ac97-codec", - .init = machine_init, .ops = NULL, }; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index be27f1d..a81cf39 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -339,8 +339,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); } - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 8fc07e9..b3a7efa 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -124,8 +124,6 @@ static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "RHPOUT"); snd_soc_dapm_nc_pin(dapm, "MICIN"); - snd_soc_dapm_sync(dapm); - return 0; } -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Convert Simtec machines to table based DAPM init
by Mark Brown 08 Oct '11

08 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/s3c24xx_simtec_hermes.c | 10 +++++----- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 10 +++++----- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 2422117..cbed1da 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -66,11 +66,6 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Line Out"); @@ -97,6 +92,11 @@ static struct snd_soc_card snd_soc_machine_simtec_aic33 = { .name = "Simtec-Hermes", .dai_link = &simtec_dai_aic33, .num_links = 1, + + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = base_map, + .num_dapm_routes = ARRAY_SIZE(base_map), }; static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 57aee5e..15961e2 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -55,11 +55,6 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Line Out"); @@ -86,6 +81,11 @@ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", .dai_link = &simtec_dai_aic23, .num_links = 1, + + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = base_map, + .num_dapm_routes = ARRAY_SIZE(base_map), }; static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) -- 1.7.6.3

1 0

[alsa-devel] [PATCH] alsa-lib - add headphone as standard pcm device
by Raymond Yau 08 Oct '11

08 Oct '11

Do you think it is feasible to add "Headphone" as a standard pcm device ? - add headphone as standard pcm device - add 'headphone' device to HDA-Intel.conf

2 1

[alsa-devel] [PATCH] ASoC: Suppress early calls to snd_soc_dapm_sync()
by Mark Brown 07 Oct '11

07 Oct '11

Ensure we only have one sync during the initial startup of the card by making snd_soc_dapm_sync() a noop on non-instantiated cards. This avoids any bounces due to things like jacks reporting their initial state on partially initialised cards. The callers that don't also get called at runtime should just be removed. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 7 +++++++ 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2a25076..b65e3d4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1498,6 +1498,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) #endif card->instantiated = 1; + snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); return; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8711aab..e49c56d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1845,6 +1845,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, */ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) { + /* + * Suppress early reports (eg, jacks syncing their state) to avoid + * silly DAPM runs during card startup. + */ + if (!dapm->card || !dapm->card->instantiated) + return 0; + return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); -- 1.7.6.3

2 2

[alsa-devel] [PATCH] ASoC: OMAP machines: Fix kernel crash due to changes in core
by Peter Ujfalusi 07 Oct '11

07 Oct '11

Calling soc_dapm_sync() after adding DAPM widgets/routes will lead to kernel crash caused by unitialized widget->power_check callback (NULL pointer dereference). Call snd_soc_dapm_new_widgets(dapm); instead of soc_dapm_sync which will initialize the widgets, and will call the dapm_power_widgets (soc_dapm_sync is a wrapper for dapm_power_widgets). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Anuj Aggarwal <anuj.aggarwal(a)ti.com> Cc: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Gražvydas Ignotas <notasas(a)gmail.com> Cc: Misael Lopez Cruz <misael.lopez(a)ti.com> --- sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap3pandora.c | 4 ++-- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 9 ++++++--- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 +- sound/soc/omap/zoom2.c | 2 +- 9 files changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 48af0f8..f9b995f 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -107,7 +107,7 @@ static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic In"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_new_widgets(dapm); return 0; } diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0aa475f..10d3d41 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -569,7 +569,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Speaker"); snd_soc_dapm_disable_pin(dapm, "AGCIN"); snd_soc_dapm_disable_pin(dapm, "AGCOUT"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_new_widgets(dapm); /* Add virtual switch */ ret = snd_soc_add_controls(codec, ams_delta_audio_controls, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c10d356..bda2aa4 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -282,7 +282,7 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) /* Set up N810 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_new_widgets(dapm); return 0; } diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 3ae87fd..35368fa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -176,7 +176,7 @@ static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, omap3pandora_out_map, ARRAY_SIZE(omap3pandora_out_map)); - return snd_soc_dapm_sync(dapm); + return snd_soc_dapm_new_widgets(dapm); } static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) @@ -199,7 +199,7 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, omap3pandora_in_map, ARRAY_SIZE(omap3pandora_in_map)); - return snd_soc_dapm_sync(dapm); + return snd_soc_dapm_new_widgets(dapm); } static struct snd_soc_ops omap3pandora_ops = { diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 5978332..4a13c45 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -107,7 +107,7 @@ static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_new_widgets(dapm); return 0; } diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 7164db5..a4676a0 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -317,7 +317,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) if (err < 0) return err; - snd_soc_dapm_sync(dapm); + snd_soc_dapm_new_widgets(dapm); /* AV jack detection */ err = snd_soc_jack_new(codec, "AV Jack", @@ -346,8 +346,11 @@ static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm) if (err < 0) return 0; - return snd_soc_dapm_add_routes(dapm, audio_mapb, - ARRAY_SIZE(audio_mapb)); + err = snd_soc_dapm_add_routes(dapm, audio_mapb, ARRAY_SIZE(audio_mapb)); + if (err < 0) + return 0; + + return snd_soc_dapm_new_widgets(dapm); } /* Digital audio interface glue - connects codec <--> CPU */ diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 269aded..4bb650f 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -159,7 +159,7 @@ static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "CARKITL"); snd_soc_dapm_nc_pin(dapm, "CARKITR"); - ret = snd_soc_dapm_sync(dapm); + ret = snd_soc_dapm_new_widgets(dapm); if (ret) return ret; diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 79ec76d..10972fb 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -140,7 +140,7 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(dapm); + ret = snd_soc_dapm_new_widgets(dapm); if (ret) return ret; diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 8b1ebbc..9e7d21d 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -126,7 +126,7 @@ static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "CARKITL"); snd_soc_dapm_nc_pin(dapm, "CARKITR"); - ret = snd_soc_dapm_sync(dapm); + ret = snd_soc_dapm_new_widgets(dapm); return ret; } -- 1.7.7

2 9

[alsa-devel] [PATCH 00/11 v2] Convert wm8xxx codec drivers to use snd_soc_write/snd_soc_cache_sync
by Axel Lin 07 Oct '11

07 Oct '11

This serial converts the wm8xxx codec drivers to use snd_soc_cache_sync. I don't have these hardware, but I assume Mark has all these hardware. These patches are UNTESTED. I'd appreciate if someone can test it. (Current code is broken anyway, thus we need convert to use new API. ) v2: For 7, 9 CODECs that are write only, convert to using cache_bypass + snd_soc_write v2 also includes wm8990 that I had sent separately. *** BLURB HERE *** Axel Lin (11): ASoC: wm8510: Convert to snd_soc_write ASoC: wm8711: Convert to snd_soc_write ASoC: wm8731: Convert to snd_soc_write ASoC: wm8750: Convert to snd_soc_write ASoC: wm8776: Convert to snd_soc_write ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_write ASoC: wm8971: Convert to snd_soc_write ASoC: wm8974: Convert to snd_soc_write ASoC: wm8988: Convert to snd_soc_write ASoC: wm8990: Convert to snd_soc_cache_sync sound/soc/codecs/wm8510.c | 33 +++++++++++++++++++++---------- sound/soc/codecs/wm8711.c | 33 +++++++++++++++++++++---------- sound/soc/codecs/wm8731.c | 35 +++++++++++++++++++++------------ sound/soc/codecs/wm8750.c | 36 +++++++++++++++++++++------------- sound/soc/codecs/wm8776.c | 36 +++++++++++++++++++++------------- sound/soc/codecs/wm8940.c | 46 +++++++++++++++++++++----------------------- sound/soc/codecs/wm8960.c | 34 +++++++++++++++++++++++--------- sound/soc/codecs/wm8971.c | 35 ++++++++++++++++++++++----------- sound/soc/codecs/wm8974.c | 33 +++++++++++++++++++++---------- sound/soc/codecs/wm8988.c | 37 +++++++++++++++++++++-------------- sound/soc/codecs/wm8990.c | 32 ++++++++++++++++++------------ 11 files changed, 242 insertions(+), 148 deletions(-) -- 1.7.4.1

2 15

[alsa-devel] [PATCH] ASoC: wm_hubs: Remove unneeded count variable in wait_for_dc_servo
by Axel Lin 07 Oct '11

07 Oct '11

The count variable is not required, decrement timeout in the while loop can achieve the same effect. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm_hubs.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4..6e2e122 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -66,7 +66,6 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; - int count = 0; int timeout; unsigned int val; @@ -83,7 +82,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) timeout = 400; do { - count++; + timeout--; if (hubs->dcs_done_irq) wait_for_completion_timeout(&hubs->dcs_done, @@ -93,9 +92,9 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) reg = snd_soc_read(codec, WM8993_DC_SERVO_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < timeout); + } while ((reg & op) && timeout); - if (reg & op) + if (timeout == 0) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", op); } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: am3517evm: Let core to deal with the DAPM widgets
by Peter Ujfalusi 07 Oct '11

07 Oct '11

Pass the DAPM widgets/routes via the snd_soc_card struct to core. With this change we do not need the init function since the remaining snd_soc_dapm_enable_pin calls are not needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/am3517evm.c | 28 +++++----------------------- 1 files changed, 5 insertions(+), 23 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index f9b995f..8da55e9 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -90,28 +90,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic In"}, }; -static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add am3517-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic In"); - - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link am3517evm_dai = { .name = "TLV320AIC23", @@ -122,7 +100,6 @@ static struct snd_soc_dai_link am3517evm_dai = { .codec_name = "tlv320aic23-codec.2-001a", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; @@ -131,6 +108,11 @@ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", .dai_link = &am3517evm_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *am3517evm_snd_device; -- 1.7.7

1 0

[alsa-devel] [PATCH] ASoC: osk5912: Let core to deal with the DAPM widgets
by Peter Ujfalusi 07 Oct '11

07 Oct '11

Pass the DAPM widgets/routes via the snd_soc_card struct to core. With this change we do not need the init function since the remaining snd_soc_dapm_enable_pin calls are not needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/osk5912.c | 27 +++++---------------------- 1 files changed, 5 insertions(+), 22 deletions(-) diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 4a13c45..db91cca 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -91,27 +91,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add osk5912 specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up osk5912 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link osk_dai = { .name = "TLV320AIC23", @@ -122,7 +101,6 @@ static struct snd_soc_dai_link osk_dai = { .codec_name = "tlv320aic23-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = osk_tlv320aic23_init, .ops = &osk_ops, }; @@ -131,6 +109,11 @@ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", .dai_link = &osk_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *osk_snd_device; -- 1.7.7

1 0

[alsa-devel] [PATCH] ASoC: n810: Let the core to register DAPM widgets/routes and controls
by Peter Ujfalusi 07 Oct '11

07 Oct '11

Pass the DAPM widgets/routes and static controls via the snd_soc_card struct to core. In this way the machine driver does not need to handle the DAPM widgets/routes. Mark the not connected pins at late_probe time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/n810.c | 47 ++++++++++++++++++----------------------------- 1 files changed, 18 insertions(+), 29 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index bda2aa4..625e886 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -253,36 +253,16 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { n810_get_input, n810_set_input), }; -static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) +static int n810_aic33_late_probe(struct snd_soc_card *card) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; - - /* Not connected */ - snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(dapm, "HPLCOM"); - snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - snd_soc_dapm_nc_pin(dapm, "MIC3L"); - snd_soc_dapm_nc_pin(dapm, "MIC3R"); - snd_soc_dapm_nc_pin(dapm, "LINE1R"); - snd_soc_dapm_nc_pin(dapm, "LINE2L"); - snd_soc_dapm_nc_pin(dapm, "LINE2R"); - - /* Add N810 specific controls */ - err = snd_soc_add_controls(codec, aic33_n810_controls, - ARRAY_SIZE(aic33_n810_controls)); - if (err < 0) - return err; - - /* Add N810 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets, - ARRAY_SIZE(aic33_dapm_widgets)); - - /* Set up N810 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_new_widgets(dapm); + snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "MIC3L"); + snd_soc_dapm_nc_pin(&card->dapm, "MIC3R"); + snd_soc_dapm_nc_pin(&card->dapm, "LINE1R"); + snd_soc_dapm_nc_pin(&card->dapm, "LINE2L"); + snd_soc_dapm_nc_pin(&card->dapm, "LINE2R"); return 0; } @@ -306,6 +286,15 @@ static struct snd_soc_card snd_soc_n810 = { .name = "N810", .dai_link = &n810_dai, .num_links = 1, + + .controls = aic33_n810_controls, + .num_controls = ARRAY_SIZE(aic33_n810_controls), + .dapm_widgets = aic33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + + .late_probe = n810_aic33_late_probe, }; static struct platform_device *n810_snd_device; -- 1.7.7

1 0

[alsa-devel] [PATCH 1/2] ASoC: sdp440: No need to call snd_soc_dapm_enable_pin at init time
by Peter Ujfalusi 07 Oct '11

07 Oct '11

All DAPM widgets are connected after they have added. Machine driver only need to mark not connected pins. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/sdp4430.c | 8 -------- 1 files changed, 0 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 10972fb..3069226 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -132,14 +132,6 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* Set up SDP4430 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* SDP4430 connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "AFML"); - snd_soc_dapm_enable_pin(dapm, "AFMR"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_new_widgets(dapm); if (ret) return ret; -- 1.7.7

1 1

[alsa-devel] [PATCH] soc: codecs: tlv320aic32x4: fix initialization of micpga routing
by Wolfram Sang 06 Oct '11

06 Oct '11

Checking the pdata-flags used 'or', so the check is always true. Use 'and' to correctly mask the flags. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Javier Martin <javier.martin(a)vista-silicon.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e93b9d1..a68982e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -685,10 +685,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } /* Mic PGA routing */ - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); } - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); } -- 1.7.2.5

2 1

[alsa-devel] [PATCH 00/10] Convert wm8xxx codec drivers to use snd_soc_cache_sync
by Axel Lin 06 Oct '11

06 Oct '11

This serial converts the wm8xxx codec drivers to use snd_soc_cache_sync. I don't have these hardware, but I assume Mark has all these hardware. These patches are UNTESTED. I'd appreciate if someone can test them. (Current code is broken anyway, thus we need convert to use new API. ) Axel Lin (10): ASoC: wm8510: Convert to snd_soc_cache_sync ASoC: wm8711: Convert to snd_soc_cache_sync ASoC: wm8731: Convert to snd_soc_cache_sync ASoC: wm8750: Convert to snd_soc_cache_sync ASoC: wm8776: Convert to snd_soc_cache_sync ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_cache_sync ASoC: wm8971: Convert to snd_soc_cache_sync ASoC: wm8974: Convert to snd_soc_cache_sync ASoC: wm8988: Convert to snd_soc_cache_sync sound/soc/codecs/wm8510.c | 30 ++++++++++++++++++---------- sound/soc/codecs/wm8711.c | 19 +++++++---------- sound/soc/codecs/wm8731.c | 29 ++++++++++++++++----------- sound/soc/codecs/wm8750.c | 32 +++++++++++++++++------------- sound/soc/codecs/wm8776.c | 33 ++++++++++++++++++------------- sound/soc/codecs/wm8940.c | 46 +++++++++++++++++++++----------------------- sound/soc/codecs/wm8960.c | 37 +++++++++++++++++++++++++---------- sound/soc/codecs/wm8971.c | 32 +++++++++++++++++++----------- sound/soc/codecs/wm8974.c | 29 +++++++++++++++++---------- sound/soc/codecs/wm8988.c | 32 +++++++++++++++++------------- 10 files changed, 185 insertions(+), 134 deletions(-) -- 1.7.4.1

3 19

[alsa-devel] [PATCH] ASoC: Ensure DAPM widgets are set up before we sync jacks
by Mark Brown 06 Oct '11

06 Oct '11

We synchronise jack state on startup - when we do that make sure that we have set up all the DAPM widgets first in case we end up touching any of the partially set up widgets when syncing the jack pins. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-jack.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 02f95aa..6c5ebd3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -189,6 +189,8 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, list_add(&(pins[i].list), &jack->pins); } + snd_soc_dapm_new_widgets(&jack->codec->card->dapm); + /* Update to reflect the last reported status; canned jack * implementations are likely to set their state before the * card has an opportunity to associate pins. -- 1.7.6.3

2 1

[alsa-devel] [PATCH v2] ASoC: Return early with -EINVAL if invalid dai format is detected
by Axel Lin 06 Oct '11

06 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Arnaud Patard <arnaud.patard(a)rtp-net.org> --- sound/soc/codecs/cs4270.c | 8 ++++---- sound/soc/codecs/cs42l51.c | 9 ++++----- 2 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5830c93..f1f237e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -261,7 +261,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int ret = 0; /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -271,7 +270,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid dai format\n"); - ret = -EINVAL; + return -EINVAL; } /* set master/slave audio interface */ @@ -284,10 +283,11 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: /* all other modes are unsupported by the hardware */ - ret = -EINVAL; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } /** diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 286878d..8c3c820 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -288,7 +288,6 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); - int ret = 0; switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -298,7 +297,7 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid DAI format\n"); - ret = -EINVAL; + return -EINVAL; } switch (format & SND_SOC_DAIFMT_MASTER_MASK) { @@ -309,11 +308,11 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, cs42l51->func = MODE_SLAVE_AUTO; break; default: - ret = -EINVAL; - break; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } struct cs42l51_ratios { -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: Replace remaining use of *_volsw_2r with *_volsw
by Peter Ujfalusi 06 Oct '11

06 Oct '11

The snd_soc_*_volsw_2r functionality has been merged to *volsw callbacks. Few places still used the get, or put variant of volsw_2r, replace those with the corresponding *_volsw. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello Stephen, Mark, It would be better to take this patch instead of the previous "ASoC: fix codec breakage caused by the volsw/volsw_2r merger" patch, so we do not need to remove the defines in a coming patch. Thank you, and again sorry about the breakage. Regards, Peter sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/codecs/wm8580.c | 8 ++++---- sound/soc/codecs/wm_hubs.c | 4 ++-- 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 50ea9d7..35f3ad8 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -355,7 +355,7 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, return 1; } - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) return ret; @@ -392,7 +392,7 @@ static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol, break; } - return snd_soc_get_volsw_2r(kcontrol, ucontrol); + return snd_soc_get_volsw(kcontrol, ucontrol); } static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 02cbf13..b256727 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -213,7 +213,7 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, reg_cache[reg] = 0; reg_cache[reg2] = 0; - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) return ret; @@ -228,15 +228,15 @@ static const struct snd_kcontrol_new wm8580_snd_controls[] = { SOC_DOUBLE_R_EXT_TLV("DAC1 Playback Volume", WM8580_DIGITAL_ATTENUATION_DACL1, WM8580_DIGITAL_ATTENUATION_DACR1, - 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), SOC_DOUBLE_R_EXT_TLV("DAC2 Playback Volume", WM8580_DIGITAL_ATTENUATION_DACL2, WM8580_DIGITAL_ATTENUATION_DACR2, - 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), SOC_DOUBLE_R_EXT_TLV("DAC3 Playback Volume", WM8580_DIGITAL_ATTENUATION_DACL3, WM8580_DIGITAL_ATTENUATION_DACR3, - 0, 0xff, 0, snd_soc_get_volsw_2r, wm8580_out_vu, dac_tlv), + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), SOC_SINGLE("DAC1 Deemphasis Switch", WM8580_DAC_CONTROL3, 0, 1, 0), SOC_SINGLE("DAC2 Deemphasis Switch", WM8580_DAC_CONTROL3, 1, 1, 0), diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f3583a5..84f33d4 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -222,7 +222,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + ret = snd_soc_put_volsw(kcontrol, ucontrol); /* Updating the analogue gains invalidates the DC servo cache */ hubs->class_w_dcs = 0; @@ -364,7 +364,7 @@ SOC_ENUM("Speaker Mode", speaker_mode), SOC_DOUBLE_R_EXT_TLV("Headphone Volume", WM8993_LEFT_OUTPUT_VOLUME, WM8993_RIGHT_OUTPUT_VOLUME, - 0, 63, 0, snd_soc_get_volsw_2r, wm8993_put_dc_servo, + 0, 63, 0, snd_soc_get_volsw, wm8993_put_dc_servo, outpga_tlv), SOC_DOUBLE_R("Headphone Switch", WM8993_LEFT_OUTPUT_VOLUME, -- 1.7.7

2 1

[alsa-devel] [PATCH] ASoC: Remove references to linux@wolfsonmicro.com
by Mark Brown 06 Oct '11

06 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 2c2a681..c387daf 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -3,7 +3,7 @@ * * Copyright 2007 Wolfson Microelectronics PLC. * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * graeme.gregory(a)wolfsonmicro.com * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 47c7fd5..7475428 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics * - * Author: Mike Arthur <linux(a)wolfsonmicro.com> + * Author: Mike Arthur <Mike.Arthur(a)wolfsonmicro.com> * * Based on wm8731.c by Richard Purdie * diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index ca646a8..8e3bfc1 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -3,7 +3,7 @@ * * Copyright 2006-2009 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <linux(a)wolfsonmicro.com> + * Author: Liam Girdwood <Liam.Girdwood(a)wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 6af23d0..08d64a6 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -3,7 +3,7 @@ * * Copyright 2007-2010 Wolfson Microelectronics PLC. * Author: Graeme Gregory - * linux(a)wolfsonmicro.com + * Graeme.Gregory(a)wolfsonmicro.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 -- 1.7.6.3

1 0

[alsa-devel] [PATCH] Add "Line In" input jack constants
by David Henningsson 06 Oct '11

06 Oct '11

Hi, Do you think this is reasonable to apply to 3.2? If so I'll continue with writing some patches that use this new constant, of course. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

5 9

[alsa-devel] [PATCH] Aplay: Fix spelling error
by David Henningsson 06 Oct '11

06 Oct '11

BugLink: https://bugs.launchpad.net/bugs/816388 Reported-by: Chad Dunlap <cldunlap1(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- aplay/aplay.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 07e6bc0..5ab63a9 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -209,7 +209,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --disable-format disable automatic format conversions\n" " --disable-softvol disable software volume control (softvol)\n" " --test-position test ring buffer position\n" -" --test-coef=# test coeficient for ring buffer position (default 8)\n" +" --test-coef=# test coefficient for ring buffer position (default 8)\n" " expression for validation is: coef * (buffer_size / 2)\n" " --test-nowait do not wait for ring buffer - eats whole CPU\n" " --max-file-time=# start another output file when the old file has recorded\n" -- 1.7.5.4

1 0

[alsa-devel] [RFC][PATCH v2] ASoC: ak4535: convert to soc-cache
by Axel Lin 06 Oct '11

06 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: Set codec->cache_sync = 1 before calling snd_soc_cache_sync(). I don't have this hardware, this patch is UNTESTED!! ( But current code is broken anyway ) I'd appreciate if someone can test it. Axel sound/soc/codecs/ak4535.c | 101 ++++++++------------------------------------ 1 files changed, 19 insertions(+), 82 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1a214e..8c497fc 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -34,7 +34,6 @@ struct ak4535_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -47,63 +46,6 @@ static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { 0x0000, 0x0000, 0x0057, 0x0000, }; -/* - * read ak4535 register cache - */ -static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4535 register cache - */ -static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the AK4535 register space - */ -static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4535 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - ak4535_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static int ak4535_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4535_CACHEREGNUM; i++) - r |= ak4535_write(codec, i, cache[i]); - - return r; -}; - static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; @@ -304,7 +246,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; if (rate) @@ -323,7 +265,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, } /* set rate */ - ak4535_write(codec, AK4535_MODE2, mode2); + snd_soc_write(codec, AK4535_MODE2, mode2); return 0; } @@ -348,44 +290,37 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, /* use 32 fs for BCLK to save power */ mode1 |= 0x4; - ak4535_write(codec, AK4535_MODE1, mode1); + snd_soc_write(codec, AK4535_MODE1, mode1); return 0; } static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); + u16 mute_reg = snd_soc_read(codec, AK4535_DAC); if (!mute) - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg & ~0x20); else - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg | 0x20); return 0; } static int ak4535_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 i, mute_reg; - switch (level) { case SND_SOC_BIAS_ON: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0); break; case SND_SOC_BIAS_PREPARE: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0x20); break; case SND_SOC_BIAS_STANDBY: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i | 0x80); - i = ak4535_read_reg_cache(codec, AK4535_PM2); - ak4535_write(codec, AK4535_PM2, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0x80); + snd_soc_update_bits(codec, AK4535_PM2, 0x80, 0); break; case SND_SOC_BIAS_OFF: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0); break; } codec->dapm.bias_level = level; @@ -428,7 +363,8 @@ static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state) static int ak4535_resume(struct snd_soc_codec *codec) { - ak4535_sync(codec); + codec->cache_sync = 1; + snd_soc_cache_sync(codec); ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -436,11 +372,15 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); + int ret; printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); - codec->control_data = ak4535->control_data; - + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -461,8 +401,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .remove = ak4535_remove, .suspend = ak4535_suspend, .resume = ak4535_resume, - .read = ak4535_read_reg_cache, - .write = ak4535_write, .set_bias_level = ak4535_set_bias_level, .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), @@ -485,7 +423,6 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4535); - ak4535->control_data = i2c; ak4535->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

1 1

[alsa-devel] [PATCH] ALSA: hdmi: fix printout of SAD sampling rates
by Pierre-Louis Bossart 06 Oct '11

06 Oct '11

SAD sampling rate information reported in /proc/asound/cardX/eldX is incorrect due to a mismatch between HDA and HDMI frequencies. Add new routine to provide relevant values. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_eld.c | 24 ++++++++++++++++++++++-- 2 files changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e3db196..34a24e4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5205,7 +5205,7 @@ EXPORT_SYMBOL_HDA(snd_array_free); * @buf: the string buffer to write * @buflen: the max buffer length * - * used by hda_proc.c and hda_eld.c + * used by hda_proc.c */ void snd_print_pcm_rates(int pcm, char *buf, int buflen) { diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index f1c621d..bc1ac294 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -363,6 +363,26 @@ error: return ret; } +/** + * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with + * hdmi-specific routine. + */ +static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) +{ + static unsigned int alsa_rates[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 384000 + }; + int i, j; + + for (i = 0, j = 0; i < ARRAY_SIZE(alsa_rates); i++) + if (pcm & (1 << i)) + j += snprintf(buf + j, buflen - j, " %d", + alsa_rates[i]); + + buf[j] = '\0'; /* necessary when j == 0 */ +} + static void hdmi_show_short_audio_desc(struct cea_sad *a) { char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; @@ -371,7 +391,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) if (!a->format) return; - snd_print_pcm_rates(a->rates, buf, sizeof(buf)); + hdmi_print_pcm_rates(a->rates, buf, sizeof(buf)); if (a->format == AUDIO_CODING_TYPE_LPCM) snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); @@ -430,7 +450,7 @@ static void hdmi_print_sad_info(int i, struct cea_sad *a, i, a->format, cea_audio_coding_type_names[a->format]); snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels); - snd_print_pcm_rates(a->rates, buf, sizeof(buf)); + hdmi_print_pcm_rates(a->rates, buf, sizeof(buf)); snd_iprintf(buffer, "sad%d_rates\t\t[0x%x]%s\n", i, a->rates, buf); if (a->format == AUDIO_CODING_TYPE_LPCM) { -- 1.7.6.2

2 1

[alsa-devel] [RFC][PATCH] ASoC: wm8990: Convert to snd_soc_cache_sync for sync reg_cache with the hardware
by Axel Lin 06 Oct '11

06 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, this patch is UNTESTED!! ( But current code is broken anyway ) I'd appreciate if someone can test it. Axel sound/soc/codecs/wm8990.c | 32 +++++++++++++++++++------------- 1 files changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 100aeee..48e9dd9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -36,6 +36,17 @@ struct wm8990_priv { unsigned int pcmclk; }; +static int wm8990_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case WM8990_RESET: + return 1; + default: + return 0; + } +} + /* * wm8990 register cache. Note that register 0 is not included in the * cache. @@ -1156,6 +1167,7 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute) static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; u16 val; switch (level) { @@ -1171,6 +1183,12 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + /* Enable all output discharge bits */ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | WM8990_DIS_RLINE | WM8990_DIS_OUT3 | @@ -1319,19 +1337,6 @@ static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8990_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { - if (i + 1 == WM8990_RESET) - continue; - data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } - wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -1392,6 +1397,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .reg_cache_size = ARRAY_SIZE(wm8990_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8990_reg, + .volatile_register = wm8990_volatile_register, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.4.1

1 1

[alsa-devel] [PATCH] ASoC: ak4535: Set codec->hw_write to i2c_master_send
by Axel Lin 06 Oct '11

06 Oct '11

It was removed in commit f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support". Add it back, otherwise codec->hw_write points to NULL. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4535.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1a214e..00cc92c 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -439,6 +439,7 @@ static int ak4535_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); + codec->hw_write = (hw_write_t)i2c_master_send; codec->control_data = ak4535->control_data; /* power on device */ -- 1.7.4.1

2 2

[alsa-devel] [PATCH v2] ASoC: Add support for cs42l73 codec
by Brian Austin 06 Oct '11

06 Oct '11

This patch adds support for the Cirrus Logic CS42L73 low power stereo codec. This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1047 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 223 ++++++++++ 4 files changed, 1276 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3449431..9107c48 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -175,6 +176,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 787881b..e89e84b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..18a49d8 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1047 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <office(a)nucleusys.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/gpio.h> + +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; + +struct cs42l73_private { + enum snd_soc_control_type control_type; + void *control_data; + u32 sysclk; + u8 mclksel; + u32 mclk; + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[] = { +/* 0*/ 0x00, 0x42, 0xA7, 0x30, +/* 4*/ 0x00, 0x00, 0xF1, 0xDF, +/* 8*/ 0x3F, 0x57, 0x53, 0x00, +/* C*/ 0x00, 0x15, 0x00, 0x15, +/*10*/ 0x00, 0x15, 0x00, 0x06, +/*14*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +/*60*/ 0x00, 0x00 +}; + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const s42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const cs42l73_mono_mixer_text[] = { + "Left", "Right", "Mono Mix"}; + +static const struct soc_enum spk_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum spk_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_DOUBLE_S8_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_S8_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0), + + SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15, 0), + + SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate Volume", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, 6, 1, + 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 1, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Attenuation Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Attenuation Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Attenuation Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Attenuation Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Attenuation Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Attenuation Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Attenuation Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Attenuation Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Attenuation Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Attenuation Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Attenuation Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Attenuation Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Attenuation Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Attenuation Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Attenuation Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Attenuation Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Attenuation Volume", + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Attenuation Volume", + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Attenuation Volume", + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Attenuation Volume", + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Attenuation Volume", + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Attenuation Volume", + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Attenuation Volume", + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Attenuation Volume", + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum), + SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum), + + SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum), + SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum), + + SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum), + SOC_ENUM("XSP Output Mixer Select", xspout_mixer_enum), + +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), + + SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_PGA("HP Amp Left", CS42L73_PWRCTL3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("HP Amp Right", CS42L73_PWRCTL3, 0, 1, NULL, 0), + + SND_SOC_DAPM_PGA("LO Amp Left", CS42L73_PWRCTL3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("LO Amp Right", CS42L73_PWRCTL3, 1, 1, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Amp", CS42L73_PWRCTL3, 2, 1, NULL, 0), + + SND_SOC_DAPM_PGA("EAR Amp", CS42L73_PWRCTL3, 3, 1, NULL, 0), + + SND_SOC_DAPM_PGA("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("LINEOUTA"), + SND_SOC_DAPM_OUTPUT("LINEOUTB"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + {"HPOUTA", NULL, "HP Amp Left"}, + {"HPOUTB", NULL, "HP Amp Right"}, + {"LINEOUTA", NULL, "LO Amp Left"}, + {"LINEOUTB", NULL, "LO Amp Right"}, + {"SPKOUT", NULL, "SPK Amp"}, + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"HP Amp Left", "DAC", "DAC Left"}, + {"HP Amp Right", "DAC", "DAC Right"}, + {"LO Amp Left", "DAC", "DAC Left"}, + {"LO Amp Right", "DAC", "DAC Right"}, + {"SPK Amp", "DAC", "DAC Left"}, + {"SPKLO Amp", "DAC", "DAC Right"}, + {"EAR Amp", "DAC", "DAC Right"}, + + {"PGA Mux Left", NULL, "LINEINA"}, + {"PGA Mux Right", NULL, "LINEINB"}, + {"PGA Mux Left", NULL, "MIC1"}, + {"PGA Mux Right", NULL, "MIC2"}, + + {"PGA Left", NULL, "PGA Mux Left"}, + {"PGA Right", NULL, "PGA Mux Right"}, + {"ADC Left", "ADC", "PGA Left"}, + {"ADC Right", "ADC", "PGA Right"}, + + {"XSPOUT", NULL, "ADC Left"}, + {"XSPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "XSPIN"}, + {"DAC Right", NULL, "XSPIN"}, + + {"ASPOUT", NULL, "ADC Left"}, + {"ASPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "ASPIN"}, + {"DAC Right", NULL, "ASPIN"}, + + {"VSPOUT", NULL, "ADC Left"}, + {"VSPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "VSPIN"}, + {"DAC Right", NULL, "VSPIN"}, +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(priv->sysclk); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) { + dev_err(codec->dev, "Invalid clk_id %u\n", clk_id); + return -EINVAL; + } + + if ((cs42l73_get_mclkx_coeff(freq) < 0)) { + dev_err(codec->dev, "Invalid sysclk %u\n", freq); + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->sysclk = freq; + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = codec_dai->id; + int inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~xSPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format is supported only in slave mode\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= xSPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & xSPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + if (format == SND_SOC_DAIFMT_DSP_B + && inv == SND_SOC_DAIFMT_IB_IF) + spc |= (xPCM_MODE0 << 4); + else + + if (format == SND_SOC_DAIFMT_DSP_B + && inv == SND_SOC_DAIFMT_IB_NF) + spc |= (xPCM_MODE1 << 4); + else + + if (format == SND_SOC_DAIFMT_DSP_A + && inv == SND_SOC_DAIFMT_IB_IF) + spc |= (xPCM_MODE1 << 4); + else + return -EINVAL; + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= xMCK_SCLK_64FS; + + } else { + /* CS42L73 Slave */ + dev_dbg(codec->dev, "DAI[%d]: Slave\n", id); + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= xMCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + cs42l73_update_asrc(codec, id, srate); + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = cs42l73->control_data; + codec->hw_write = (hw_write_t)i2c_master_send; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l73->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = snd_soc_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = snd_soc_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = snd_soc_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = snd_soc_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + + snd_soc_add_controls(codec, cs42l73_snd_controls, + ARRAY_SIZE(cs42l73_snd_controls)); + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = ARRAY_SIZE(cs42l73_reg), + .reg_cache_default = cs42l73_reg, + .reg_word_size = sizeof(u8), + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc(sizeof(struct cs42l73_private), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + cs42l73->control_data = i2c_client; + cs42l73->control_type = SND_SOC_I2C; + + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <office(a)nucleusys.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..aa09c2e --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,223 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev <office(a)nucleusys.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define xSP_3ST (1 << 7) +#define xSPDIF_I2S 0 +#define xSPDIF_PCM (1 << 6) +#define xPCM_MODE0 0 +#define xPCM_MODE1 1 +#define xPCM_MODE2 2 +#define xPCM_BO_MSBLSB 0 +#define xPCM_BO_LSBMSB 1 +#define xMCK_SCLK_64FS 0 +#define xMCK_SCLK_MCLK 2 +#define xMCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Mute */ +#define HPMUTE (1 << 7) +/* LO A/B Mute */ +#define LOMUTE (1 << 7) +/* SPK Digital Mute */ +#define SPKDMUTE (1 << 2) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Remove unused variable 'wm9090' in wm9090_probe
by Axel Lin 05 Oct '11

05 Oct '11

Eliminate below build warning: CC sound/soc/codecs/wm9090.o sound/soc/codecs/wm9090.c: In function 'wm9090_probe': sound/soc/codecs/wm9090.c:550: warning: unused variable 'wm9090' Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9090.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 228d782..2b5252c 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -548,7 +548,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { - struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); int ret; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Remove unused function declaration in imx-ssi.h
by Axel Lin 05 Oct '11

05 Oct '11

These functions are removed in commit f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support". Let's remove the leftover function declaration in header file. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/imx-ssi.h | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index 0a84cec..1072dfb 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -218,12 +218,6 @@ struct imx_ssi { struct platform_device *soc_platform_pdev_fiq; }; -struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, - struct imx_ssi *ssi); -void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); -struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, - struct imx_ssi *ssi); - int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); void imx_pcm_free(struct snd_pcm *pcm); -- 1.7.4.1

2 2

[alsa-devel] [PATCH] ASoC: Remove impossible case from wm8994_hw_params
by Axel Lin 05 Oct '11

05 Oct '11

We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params. Thus no need to check wm8994-aif3 in wm8994_hw_params. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8994.c | 9 --------- 1 files changed, 0 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d5b69c0..94a8db2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2210,7 +2210,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; int aif2_reg; @@ -2253,14 +2252,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "AIF2 using split LRCLK\n"); } break; - case 3: - switch (control->type) { - case WM8958: - aif1_reg = WM8958_AIF3_CONTROL_1; - break; - default: - return 0; - } default: return -EINVAL; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Return early with -EINVAL if invalid dai format is detected
by Axel Lin 05 Oct '11

05 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Arnaud Patard <arnaud.patard(a)rtp-net.org> --- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5830c93..e645d67 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -271,7 +271,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid dai format\n"); - ret = -EINVAL; + return -EINVAL; } /* set master/slave audio interface */ diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 286878d..779294b 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -298,7 +298,7 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid DAI format\n"); - ret = -EINVAL; + return -EINVAL; } switch (format & SND_SOC_DAIFMT_MASTER_MASK) { -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2 00/10] Combining snd_soc_*_volsw/volsw_2r
by Peter Ujfalusi 05 Oct '11

05 Oct '11

Hello Mark, Changes since v1 (RFC series): - rebased on git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch - twl6040: warning added for the custom volsw calls in case of unexpected parameter - snd_soc_volsw_is_stereo returns type changed to bool - patch 06: optimized the if() cases in order to reduce the number of comparison in certain cases - patch 06: Use the (reg == reg2) comparison instead of (shift != rshitf) to differentiate between single or 2r type of stereo controls - patch 07: Use the (reg == reg2) comparison instead of (shift != rshitf) to differentiate between single or 2r type of stereo controls Intro mail from v1: This series will combine the snd_soc_*_volsw, and snd_soc_*_volsw_2r callbacks. We are going to drop the volsw_r2 callbacks, and merge the support to the _volsw functions. In order to achieve this, the macros need to be adjusted, and also the drivers, which were not using the macros to construct the soc_mixer_control needed to be changed, so I'm not going to introduce any breakage, inconsistency. This has been suggested in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044287.… By Mark in this mail (I hope I did not misunderstood it): http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044343.… The use of the existing macros will not change, but underneath we are not going to have two sets of functions for single, or double mixer controls. I think the last 3 patch can be squashed, I did not wanted to touch the core, and a codec driver in a same patch. Regards, Peter --- Peter Ujfalusi (10): ASoC: Consolidate use of controls with custom get/put function ASoC: twl6040: Simplify custom get_volsw callback ASoC: twl6040: Prepare for core put_volsw/volsw_2r merger ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions ASoC: core: Combine snd_soc_get_volsw/get_volsw_2r functions ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions ASoC: twl6040: Simply call snd_soc_put_volsw form the custom code ASoC: core: Remove snd_soc_put_volsw_2r definition ASoC: twl6040: Warn user in twl6040_put_volsw for error case include/sound/soc.h | 33 +++++--- sound/soc/codecs/tlv320aic23.c | 14 +--- sound/soc/codecs/twl4030.c | 48 +++--------- sound/soc/codecs/twl6040.c | 64 ++++------------ sound/soc/codecs/wm8350.c | 39 ++++------ sound/soc/codecs/wm8580.c | 36 +++------ sound/soc/codecs/wm_hubs.c | 18 +--- sound/soc/soc-core.c | 162 +++++++++------------------------------- 8 files changed, 117 insertions(+), 297 deletions(-) -- 1.7.7

2 11

[PATCH] Add "Line In" input jack constants
by David Henningsson 05 Oct '11

05 Oct '11

Similar to Line Out, these constants form the base for future patches enabling input jack reporting for Line in jacks. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/linux/input.h | 1 + include/sound/jack.h | 1 + sound/core/jack.c | 1 + sound/pci/hda/hda_codec.c | 2 ++ 4 files changed, 5 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index a637e78..a514fb8 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -814,6 +814,7 @@ struct input_keymap_entry { #define SW_KEYPAD_SLIDE 0x0a /* set = keypad slide out */ #define SW_FRONT_PROXIMITY 0x0b /* set = front proximity sensor active */ #define SW_ROTATE_LOCK 0x0c /* set = rotate locked/disabled */ +#define SW_LINEIN_INSERT 0x0d /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index c140fc7..63c7907 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -42,6 +42,7 @@ enum snd_jack_types { SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ SND_JACK_VIDEOOUT = 0x0010, SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT, + SND_JACK_LINEIN = 0x0020, /* Kept separate from switches to facilitate implementation */ SND_JACK_BTN_0 = 0x4000, diff --git a/sound/core/jack.c b/sound/core/jack.c index 53b53e9..240a3e1 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -30,6 +30,7 @@ static int jack_switch_types[] = { SW_LINEOUT_INSERT, SW_JACK_PHYSICAL_INSERT, SW_VIDEOOUT_INSERT, + SW_LINEIN_INSERT, }; static int snd_jack_dev_free(struct snd_device *device) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e3db196..8b046a1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5264,6 +5264,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, return "Mic"; case SND_JACK_LINEOUT: return "Line-out"; + case SND_JACK_LINEIN: + return "Line-in"; case SND_JACK_HEADSET: return "Headset"; case SND_JACK_VIDEOOUT: -- 1.7.5.4 --------------050706010401040202030702--

1 0

[alsa-devel] [PATCH] ASoC: Use dapm_mark_dirty() for new DAPM widgets for consistency
by Mark Brown 05 Oct '11

05 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 82d93bf..8711aab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2185,7 +2185,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->new = 1; - list_add(&w->dirty, &(w->dapm->card->dapm_dirty)); + dapm_mark_dirty(w, "new widget"); dapm_debugfs_add_widget(w); } -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ALSA: HDA: Fix DAC assignment for secondary headphone on Sigmatel/IDT
by David Henningsson 05 Oct '11

05 Oct '11

This patch should be safe enough for 3.2, but do you think it is safe enough for stable as well? If we run out of DACs when trying to assign a DAC to a secondary headphone, prefer the DAC of the first headphone to the primary (usually line out) DAC. BugLink: http://bugs.launchpad.net/bugs/845275 (there are codec info at this link as well) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [RFC][PATCH] ASoC: Fix sync reg_cache with the hardware in wm8990_resume
by Axel Lin 05 Oct '11

05 Oct '11

Current code has off-by-one bug of checking WM8990_RESET. Besides, according to wm8990 datasheet (2-wire serial control mode): A control word consists of 24 bits. The first 8 bits are address bits, the remaining 16 bits are data bits. Current code uses 7 bits address bits and 9 bits data bits which is incorrect. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I'd appreciate if someone can test it. I'm a little bit confused if this fix is correct, why there is no bug report so far. If I'm wrong, please kindly let me know. Thanks, Axel sound/soc/codecs/wm8990.c | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 100aeee..5546f5a 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1320,16 +1320,17 @@ static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8990_resume(struct snd_soc_codec *codec) { int i; - u8 data[2]; + u8 data[3]; u16 *cache = codec->reg_cache; /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { - if (i + 1 == WM8990_RESET) + if (i == WM8990_RESET) continue; - data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); + data[0] = i; + data[1] = (cache[i] & 0xFF00) >> 8; + data[2] = cache[i] & 0x00FF; + codec->hw_write(codec->control_data, data, 3); } wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- 1.7.4.1

2 2

[PATCH] ALSA: HDA: Fix DAC assignment for secondary headphone on Sigmatel/IDT
by David Henningsson 05 Oct '11

05 Oct '11

If we run out of DACs when trying to assign a DAC to a secondary headphone, prefer the DAC of the first headphone to the primary (usually line out) DAC. BugLink: http://bugs.launchpad.net/bugs/845275 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dd977b6..1e0b338 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2972,8 +2972,9 @@ static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int j, conn_len; - hda_nid_t conn[HDA_MAX_CONNECTIONS]; + hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac; unsigned int wcaps, wtype; conn_len = snd_hda_get_connections(codec, nid, conn, @@ -3001,10 +3002,21 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) return conn[j]; } } - /* if all DACs are already assigned, connect to the primary DAC */ + + /* if all DACs are already assigned, connect to the primary DAC, + unless we're assigning a secondary headphone */ + fallback_dac = spec->multiout.dac_nids[0]; + if (spec->multiout.hp_nid) { + for (j = 0; j < cfg->hp_outs; j++) + if (cfg->hp_pins[j] == nid) { + fallback_dac = spec->multiout.hp_nid; + break; + } + } + if (conn_len > 1) { for (j = 0; j < conn_len; j++) { - if (conn[j] == spec->multiout.dac_nids[0]) { + if (conn[j] == fallback_dac) { snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, j); break; -- 1.7.5.4 --------------040409070004050505030701--

1 0

[alsa-devel] [PATCH 1/4] ASoC: Add verbose debugging showing why widgets get marked dirty
by Mark Brown 05 Oct '11

05 Oct '11

Help diagnose why we're checking widgets by providing some logging when we first dirty them. This should possibly be a trace point if it's useful but can be absurdly verbose if enabled, we can always change it later if desired. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 26 +++++++++++++++----------- 1 files changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 214a709..e6a0882 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,10 +124,13 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -static void dapm_mark_dirty(struct snd_soc_dapm_widget *w) +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { - if (!dapm_dirty_widget(w)) + if (!dapm_dirty_widget(w)) { + dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", + w->name, reason); list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); + } } /* create a new dapm widget */ @@ -1227,7 +1230,7 @@ static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, /* If the peer is already in the state we're moving to then we * won't have an impact on it. */ if (power != peer->power) - dapm_mark_dirty(peer); + dapm_mark_dirty(peer, "peer state change"); } static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, @@ -1624,16 +1627,17 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, /* we now need to match the string in the enum to the path */ if (!(strcmp(path->name, e->texts[mux]))) { path->connect = 1; /* new connection */ - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, "mux connection"); } else { if (path->connect) - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, + "mux disconnection"); path->connect = 0; /* old connection must be powered down */ } } if (found) { - dapm_mark_dirty(widget); + dapm_mark_dirty(widget, "mux change"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } @@ -1660,11 +1664,11 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, "mixer connection"); } if (found) { - dapm_mark_dirty(widget); + dapm_mark_dirty(widget, "mixer update"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } @@ -1810,7 +1814,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, w->connected = status; if (status == 0) w->force = 0; - dapm_mark_dirty(w); + dapm_mark_dirty(w, "pin configuration"); return 0; } @@ -2699,7 +2703,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { - dapm_mark_dirty(w); + dapm_mark_dirty(w, "stream event"); switch(event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; @@ -2789,7 +2793,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, dev_dbg(w->dapm->dev, "dapm: force enable pin %s\n", pin); w->connected = 1; w->force = 1; - dapm_mark_dirty(w); + dapm_mark_dirty(w, "force enable"); return 0; } -- 1.7.6.3

1 3

[alsa-devel] Future of the HDA driver
by David Henningsson 04 Oct '11

04 Oct '11

Hi Takashi etc, 1) I think it would make sense to have a designated time and room for a "future of HDA" discussion in Prague. We could e g discuss input jacks as kcontrols, and exposing routing to user space, as IIRC Mark Brown was suggestion earlier. What do you think? 2) With Ubuntu 11.10 in a "Freeze" state and PulseAudio 1.0 out the door, I might have some time to contribute to the HDA driver...at least if not a lot of urgent stuff comes up, and up to the 3.2 merge window or so. (Any idea how far away that would be?) Do you think it would make sense to split hda_codec.c into hda_codec.c and hda_autoparser.c, move snd_hda_parse_pin_defcfg there, then add more functions as discovered to be useful to more than one autoparser? 3) One thing that has been annoyed me lately is the moving of hp out or speaker out to line out, which IMO leads to somewhat messy code. Seen in retrospect of course, don't you think it would make more sense to do let line_out_pins be "line out" only, and add one more variable primary_out that would be initialized as: primary_out_pins = line_outs ? &line_out_pins : (speaker_outs ? &speaker_out_pins : (hp_outs ? &hp_out_pins : NULL)) ..and then you could do something like: #define hp_is_primary_out(_auto) ((_auto)->primary_out_pins == &(_auto)->hp_out_pins) Redoing that will probably require careful reading to avoid regressions though... -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 13

[alsa-devel] [RFC 00/12] Combining snd_soc_*_volsw/volsw_2r
by Peter Ujfalusi 04 Oct '11

04 Oct '11

Hello Mark, This series will combine the snd_soc_*_volsw, and snd_soc_*_volsw_2r callbacks. We are going to drop the volsw_r2 callbacks, and merge the support to the _volsw functions. In order to achieve this, the macros need to be adjusted, and also the drivers, which were not using the macros to construct the soc_mixer_control needed to be changed, so I'm not going to introduce any breakage, inconsistency. This has been suggested in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044287.… By Mark in this mail (I hope I did not misunderstood it): http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044343.… The use of the existing macros will not change, but underneath we are not going to have two sets of functions for single, or double mixer controls. I think the last 3 patch can be squashed, I did not wanted to touch the core, and a codec driver in a same patch. Generated on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch Regards, Peter --- Peter Ujfalusi (12): ASoC: core: Introduce SOC_DOUBLE_VALUE macro ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro ASoC: Consolidate use of controls with custom get/put function ASoC: twl6040: Simplify custom get_volsw callback ASoC: twl6040: Simplify custom put_volsw callback ASoC: twl6040: Prepare for core put_volsw/volsw_2r merger ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions ASoC: core: Combine snd_soc_get_volsw/get_volsw_2r functions ASoC: core: Combine snd_soc_out_volsw/put_volsw_2r functions ASoC: twl6040: Simply call snd_soc_put_volsw form the custom code ASoC: core: Remove snd_soc_put_volsw_2r definition include/sound/soc.h | 82 +++++++++++---------- sound/soc/codecs/tlv320aic23.c | 14 +--- sound/soc/codecs/twl4030.c | 48 +++--------- sound/soc/codecs/twl6040.c | 71 ++++-------------- sound/soc/codecs/wm8350.c | 39 ++++------ sound/soc/codecs/wm8580.c | 36 +++------ sound/soc/codecs/wm_hubs.c | 18 +--- sound/soc/soc-core.c | 161 +++++++++------------------------------- 8 files changed, 142 insertions(+), 327 deletions(-) -- 1.7.7

3 21

[alsa-devel] [PATCH 1/2] ASoC: Remove unneeded hw_write initialisation in ak4671
by Axel Lin 04 Oct '11

04 Oct '11

It is not required now. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4671.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 2ecf128..41e3d55 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -628,8 +628,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec); int ret; - codec->hw_write = (hw_write_t)i2c_master_send; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- 1.7.4.1

2 2

[alsa-devel] [PATCH 1/4 v2] ASoC: wm8711: Use snd_soc_update_bits for read-modify-write
by Axel Lin 04 Oct '11

04 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: also remove unused variable 'reg'. sound/soc/codecs/wm8711.c | 8 +++----- 1 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 8457d3c..47c7fd5 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -364,7 +364,7 @@ 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, reg; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type); if (ret < 0) { @@ -381,10 +381,8 @@ static int wm8711_probe(struct snd_soc_codec *codec) wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = snd_soc_read(codec, WM8711_LOUT1V); - snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8711_ROUT1V); - snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100); + snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8711_snd_controls, ARRAY_SIZE(wm8711_snd_controls)); -- 1.7.4.1

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[alsa-devel] [RFC][PATCH] ASoC: Avoid writing to WM8971_RESET in wm8971_resume
by Axel Lin 04 Oct '11

04 Oct '11

Writing to WM8971_RESET resets all registers to the default state. Thus we should avoid writing to WM8971_RESET on resume. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, While reading the code, I don't understand why we need to do (in wm8971_resume): for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) { if (i + 1 == WM8971_RESET) continue; My understanding is that we want to avoid writing to WM8971_RESET on resume. Thus I send this patch for review. I don't have the hardware for testing, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/wm8971.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 572bb80..7ec4165 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -612,7 +612,7 @@ static int wm8971_resume(struct snd_soc_codec *codec) /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) { - if (i + 1 == WM8971_RESET) + if (i == WM8971_RESET) continue; data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; -- 1.7.4.1

2 1

[alsa-devel] ASoC updates for 3.2
by Mark Brown 04 Oct '11

04 Oct '11

The following changes since commit 02890535269338a6d2034ad3ce8b22beb24b449a: ASoC: ssm2602: Support setting the oscillator and the clock output state (2011-09-27 13:30:48 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 One MFD change for wm1811 which is here to avoid collisions with wm1811. Arnd Bergmann (5): ASoC: use a valid device for dev_err() in Zylonite ASoC: omap_mcpdm_remove cannot be __devexit ASoC: imx: eukrea_tlv320 needs i2c ASoC: sh: use correct __iomem annotations ASoC: samsung: WM8994 depends on MFD_WM8994 Axel Lin (41): ASoC: Remove unused "control_data" field of struct wm8940_priv ASoC: Remove unused "control_data" field of struct wm8960_priv ASoC: Remove unused "control_data" field of struct wm8978_priv ASoC: Remove unused "control_data" field of struct wm9081_priv ASoC: Remove unused "control_data" field of struct wm9090_priv ASoC: Remove unused "control_data" field of struct wm8904_priv ASoC: Fix setting adau1373_dai->master for SND_SOC_DAIFMT_CBS_CFS ASoC: adau1701: Initialize codec->control_data before using it ASoC: Remove unused "control_data" field of struct alc5623_priv ASoC: Remove unused "control_data" field of struct cs4270_private ASoC: Remove unused "control_data" field of struct cs42l51_private ASoC: Remove unused "control_data" field of struct max98088_priv ASoC: Remove unused "control_data" field of struct max98095_priv ASoC: Remove unused "control_data" field of struct ak4671_priv ASoC: adau1701: Fix prototype for adau1701_set_sysclk ASoC: tlv320dac33: Add guarding parentheses to macros ASoC: Remove unused "control_data" field of struct cs4271_private ASoC: Remove unused "control_data" field of struct aic3x_priv ASoC: Add missed regulator_unregister_notifier and regulator_bulk_free in wm8995_remove ASoC: sn95031: Do not use static variable for channel_index ASoC: tlv320aic3x: Use driver_data field of struct i2c_device_id to identify models ASoC: Fix setting update bits for WM8753_LADC and WM8753_RADC ASoC: wm8782: Add __devexit_p at necessary place ASoC: kirkwood-i2s: Add __devexit_p at necessary place ASoC: samsung: Add __devexit_p at necessary places ASoC: tegra: Staticise tegra_i2s_dai and tegra_spdif_dai ASoC: Staticise samsung_spdif_dai ASoC: Staticise sh4_ssi_dai ASoC: Remove unused rate variable in magician_playback_hw_params ASoC: Remove unused srate variable in tegra_spdif_hw_params ASoC: Remove needless codec->dapm.bias_level assignment to SND_SOC_BIAS_OFF ASoC: Staticise simtec_audio_resume() ASoC: Staticise ep93xx_ac97_dai ASoC: txx9: Add __exit_p at necessary place ASoC: Fix setting update bits for WM8741_DACRMSB_ATTENUATION ASoC: samsung: s3c-i2s-v2.c needs module.h ASoC: wm8711: Use snd_soc_update_bits for read-modify-write ASoC: wm8971: Use snd_soc_update_bits for read-modify-write ASoC: wm8988: Use snd_soc_update_bits for read-modify-write ASoC: wm8994: Slightly optimize configure_clock ASoC: wm8995: Slightly optimize configure_clock Jarkko Nikula (5): ASoC: Davinci: Fix FS polarity for I2S format ASoC: omap-mcbsp: Fix FS polarity for LEFT_J, DSP_A and DSP_B formats ASoC: omap-mcbsp: Prepare for init time DAI format setting ASoC: omap: Convert bunch of machine drivers to use init time DAI format ASoC: omap: Use single hw_params callback in sdp3430 and zoom2 Karl Tsou (1): ASoC: Add DRC control for WM8996 Mark Brown (12): ASoC: Add device ID for WM9093 to WM9090 driver ASoC: Allow DAI formats to be specified in the dai_link ASoC: Use dai_fmt in speyside_wm8962 ASoC: Also count neighbour checks for supplies ASoC: Staticise non-exported symbols in rt5631 ASoC: Remove unused function check_vdac_to_outmix from rt5631 Merge branch 'for-3.1' into for-3.2 Merge branch 'for-3.1' into for-3.2 ASoC: Instantiate DAPM widgets before we do the DAI link init ASoC: Fix typo in 24.576MHz rate in WM5100 mfd: Add WM1811 support ASoC: Add WM1811 support Peter Ujfalusi (5): ASoC: twl6040: One workqueue should be enough ASoC: twl6040: correct loop counters for HS/HF ramp code ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event ASoC: twl6040: Simplify code in out_drv_event for pending work check ASoC: omap-pcm: Fix the no period wakeup implementation Ryan Mallon (2): ASoC: max98088 codec: Catch driver bugs for eq channel name ASoC: max98095 codec: Catch driver bugs for biquad channel name Susan Gao (1): ASoC: Fix a bug in WM8962 DSP_A and DSP_B settings Ujfalusi, Peter (1): ASoC: sdp4430: Fix string for FM input name drivers/mfd/wm8994-core.c | 27 ++++++++++ include/linux/mfd/wm8994/core.h | 1 + include/linux/mfd/wm8994/registers.h | 4 ++ include/sound/soc-dai.h | 34 ++++++------ include/sound/soc.h | 2 + sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 3 +- sound/soc/codecs/ak4671.c | 2 - sound/soc/codecs/alc5623.c | 2 - sound/soc/codecs/cs4270.c | 6 +-- sound/soc/codecs/cs4271.c | 5 -- sound/soc/codecs/cs42l51.c | 6 +-- sound/soc/codecs/max98088.c | 36 ++++++++----- sound/soc/codecs/max98095.c | 34 ++++++++---- sound/soc/codecs/rt5631.c | 20 +------ sound/soc/codecs/sn95031.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 16 ++---- sound/soc/codecs/tlv320dac33.c | 6 +- sound/soc/codecs/twl6040.c | 81 ++++++++++++------------------ sound/soc/codecs/wm5100.c | 6 +-- sound/soc/codecs/wm8711.c | 6 +-- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8782.c | 2 +- sound/soc/codecs/wm8904.c | 2 - sound/soc/codecs/wm8940.c | 3 - sound/soc/codecs/wm8960.c | 3 - sound/soc/codecs/wm8962.c | 4 +- sound/soc/codecs/wm8971.c | 27 +++------- sound/soc/codecs/wm8978.c | 3 - sound/soc/codecs/wm8988.c | 15 ++---- sound/soc/codecs/wm8994.c | 94 +++++++++++++++++++++++++++++---- sound/soc/codecs/wm8995.c | 19 ++++--- sound/soc/codecs/wm8996.c | 9 +++- sound/soc/codecs/wm9081.c | 3 - sound/soc/codecs/wm9090.c | 4 +- sound/soc/davinci/davinci-i2s.c | 5 ++- sound/soc/ep93xx/ep93xx-ac97.c | 2 +- sound/soc/imx/Kconfig | 1 + sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/omap/am3517evm.c | 22 +------- sound/soc/omap/igep0020.c | 23 +-------- sound/soc/omap/n810.c | 19 +------ sound/soc/omap/omap-mcbsp.c | 21 ++++++-- sound/soc/omap/omap-pcm.c | 8 +++ sound/soc/omap/omap3evm.c | 23 +-------- sound/soc/omap/omap3pandora.c | 20 ++------ sound/soc/omap/osk5912.c | 23 +-------- sound/soc/omap/overo.c | 23 +-------- sound/soc/omap/rx51.c | 20 +------- sound/soc/omap/sdp3430.c | 70 ++----------------------- sound/soc/omap/sdp4430.c | 6 +- sound/soc/omap/zoom2.c | 70 ++----------------------- sound/soc/pxa/magician.c | 3 +- sound/soc/pxa/zylonite.c | 8 ++-- sound/soc/samsung/Kconfig | 6 ++ sound/soc/samsung/ac97.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 1 + sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/spdif.c | 4 +- sound/soc/samsung/speyside_wm8962.c | 19 +------ sound/soc/sh/fsi.c | 10 ++-- sound/soc/sh/ssi.c | 2 +- sound/soc/soc-core.c | 27 +++++++++- sound/soc/soc-dapm.c | 2 + sound/soc/tegra/tegra_i2s.c | 2 +- sound/soc/tegra/tegra_spdif.c | 5 +- sound/soc/txx9/txx9aclc-generic.c | 2 +- 72 files changed, 389 insertions(+), 568 deletions(-)

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[alsa-devel] ASoC: wm8994: Is the rate_reg for wm8994-aif2 correct?
by Axel Lin 04 Oct '11

04 Oct '11

hi Mark, While reading wm8994.c, it seems the correct rate_reg for wm8994-aif2 should be WM8994_AIF2_RATE. I cannot find wm8994 datasheet, could you help checking it? Thanks, Axel

2 1

[alsa-devel] ASoC updates for 3.1
by Mark Brown 04 Oct '11

04 Oct '11

The following changes since commit 9058020cd9ae3423d6fe7de591698dc96b6701aa: ASoC: ssm2602: Re-enable oscillator after suspend (2011-09-27 11:20:38 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.1 None of these should be super critical so no need to rush them out but if there's an opportunity to send stuff they're small, local fixes which are nice to have. Arnd Bergmann (2): ASoC: use a valid device for dev_err() in Zylonite ASoC: omap_mcpdm_remove cannot be __devexit Axel Lin (1): ASoC: Fix setting update bits for WM8753_LADC and WM8753_RADC sound/soc/codecs/wm8753.c | 4 ++-- sound/soc/omap/mcpdm.c | 2 +- sound/soc/omap/mcpdm.h | 2 +- sound/soc/pxa/zylonite.c | 8 ++++---- 4 files changed, 8 insertions(+), 8 deletions(-)

2 3

[PATCH] ASoC: wm8994: Fix setting rate_reg for wm8994-aif2
by Axel Lin 04 Oct '11

04 Oct '11

For wm8994-aif2, the rate_reg should be WM8994_AIF2_RATE. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4225e1f..d5b69c0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2421,7 +2421,7 @@ static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, rate_reg = WM8994_AIF1_RATE; break; case 2: - rate_reg = WM8994_AIF1_RATE; + rate_reg = WM8994_AIF2_RATE; break; default: break; -- 1.7.4.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8994: Slightly optimize configure_clock
by Axel Lin 04 Oct '11

04 Oct '11

snd_soc_update_bits() will only write new register value if the old value is different from the new value. In additional, snd_soc_update_bits() returns 0 for no change. No need to read WM8994_CLOCKING_1 register before calling snd_soc_update_bits(). Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8994.c | 11 ++++------- 1 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e537267..4225e1f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -208,7 +208,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) static int configure_clock(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int old, new; + int change, new; /* Bring up the AIF clocks first */ configure_aif_clock(codec, 0); @@ -229,14 +229,11 @@ static int configure_clock(struct snd_soc_codec *codec) else new = 0; - old = snd_soc_read(codec, WM8994_CLOCKING_1) & WM8994_SYSCLK_SRC; - - /* If there's no change then we're done. */ - if (old == new) + change = snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSCLK_SRC, new); + if (!change) return 0; - snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new); - snd_soc_dapm_sync(&codec->dapm); return 0; -- 1.7.4.1

2 2

[alsa-devel] [RFC 0/3] ASoC: davinci- pcm and McASP error detection
by Ben Gardiner 04 Oct '11

04 Oct '11

Both the EDMA3 and McASP have the ability to report error conditions. In both cases these errors are silently dropped on the floor. I would like very much to get HW errors up the stack to the aplay process using the harware. This is an RFC series demonstrating a first attempt. 1/3 isn't nearly as half-baked as the others. It proposes to return SNDRV_PCM_POS_XRUN from davinci-pcm's pointer() function like is currently done in fsl_dma. The problem I have observed in test is that the stream does not abort; several underruns are reported before aplay finally dies. I would prefer immeadiate death with a reason. 2/3 and 3/3 are a first-strike at registering 'health' callbacks provided by the CPU DAI so that error conditions can be reported back. The health callback is polled on calls to pointer(). This may be too often. Also the current method of registering these callbacks seems wrong -- please help me out here. If there is an established method of bubbling HW errors up to userspace that I completely missed: I am sorry for the noise and would be glad to hear about how to use it with davinci-pcm. Best Regards, Ben Gardiner Ben Gardiner (3): [RFC] ASoC: davinci-pcm: latch EDMA errors [RFC] ASoC: davinci-pcm: add cpu-dai health callbacks [RFC] ASoC: davinci-mcasp: add cpu dai health callback sound/soc/davinci/davinci-mcasp.c | 26 ++++++++++++++++++++++++++ sound/soc/davinci/davinci-pcm.c | 19 +++++++++++++++++-- sound/soc/davinci/davinci-pcm.h | 3 +++ 3 files changed, 46 insertions(+), 2 deletions(-) -- 1.7.4.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: max98088 codec: Catch driver bugs for eq channel name
by Mark Brown 04 Oct '11

04 Oct '11

On Tue, Oct 04, 2011 at 09:55:40AM +1100, Ryan Mallon wrote: > Move the EQ channel names to a separate array and iterate over it in > max98088_get_channel rather than duplicating the hardcoded channel > names. Add an error message if an invalid channel is passed and check > the error in the callers. Applied both, thanks.

1 0

[alsa-devel] [PATCH 1/4] ASoC: wm8711: Use snd_soc_update_bits for read-modify-write
by Axel Lin 04 Oct '11

04 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8711.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 8457d3c..6ecf1ab 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -381,10 +381,8 @@ static int wm8711_probe(struct snd_soc_codec *codec) wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = snd_soc_read(codec, WM8711_LOUT1V); - snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100); - reg = snd_soc_read(codec, WM8711_ROUT1V); - snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100); + snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100); snd_soc_add_controls(codec, wm8711_snd_controls, ARRAY_SIZE(wm8711_snd_controls)); -- 1.7.4.1

2 4

[alsa-devel] [PATCH] ASoC: omap-pcm: Fix the no period wakeup implementation
by Peter Ujfalusi 04 Oct '11

04 Oct '11

After omap_request_dma the BLOCK_IRQ is enabled as default configuration for the channel. If we are requested for no period wakeup, we need to disable the BLOCK_IRQ in order to not receive any interrupts. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-pcm.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9b5c88a..5e37ec9 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -198,6 +198,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else if (!substream->runtime->no_period_wakeup) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + else { + /* + * No period wakeup: + * we need to disable BLOCK_IRQ, which is enabled by the omap + * dma core at request dma time. + */ + omap_disable_dma_irq(prtd->dma_ch, OMAP_DMA_BLOCK_IRQ); + } if (!(cpu_class_is_omap1())) { omap_set_dma_src_burst_mode(prtd->dma_ch, -- 1.7.7

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Re: [alsa-devel] [PATCH 1/3] mfd: Add WM1811 support
by Mark Brown 04 Oct '11

04 Oct '11

On Tue, Oct 04, 2011 at 12:19:02PM +0200, Samuel Ortiz wrote: > On Tue, Sep 27, 2011 at 07:50:05PM +0100, Mark Brown wrote: > > Due to the ASoC cross dependencies it seems like it'll be easier to > > merge this via ASoC - Samuel, does this make sense to you? > It does. > Liam, please add my: > Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> Hrm, actually there's a bit of overlap with "mfd: Add more checks for WM8994 being active during suspend" from your tree. It's only a one line change (adding the WM1811 device ID) so I don't see any problem with just applying that during -rc, or I could cherry-pick that change over now? Anyway, I've applied the patch just now.

1 0

[alsa-devel] ASoC: snd_soc_codec_set_cache_io does not set codec->hw_write callback now
by Axel Lin 04 Oct '11

04 Oct '11

Hi, I just found that after commit be3ea3b9e "SoC: Use new register map API for ASoC generic physical I/O", snd_soc_codec_set_cache_io does not set codec->hw_write any more. It is still used in some drivers. e.g. In wm8988_resume, we call codec->hw_write(codec->control_data, data, 2); For the drivers that is using snd_soc_codec_set_cache_io now, where is the place to set codec->hw_write callback now? Regards, Axel

2 1

[alsa-devel] [patch] ALSA: oss-mixer - use strlcpy() instead strcpy()
by Dan Carpenter 04 Oct '11

04 Oct '11

This is mostly a static checker fix more than anything else. We're copying from a 64 char buffer into a 44 char buffer. The 64 character buffer is str[] in snd_mixer_oss_build_test_all(). The call tree is: snd_mixer_oss_build_test_all() -> snd_mixer_oss_build_test() -> snd_mixer_oss_build_test(). We never actually do fill str[] buffer all the way to 64 characters. The longest string is: sprintf(str, "%s Playback Switch", ptr->name); ptr->name is a 32 character buffer so 32 plus 16 characters for " Playback Switch" still puts us over the 44 limit from "id.name". Most likely ptr->name never gets filled to the limit, but we can't really change the size of that buffer so lets just use strlcpy() here and be safe. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 27ac611..18297f7 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -500,7 +500,7 @@ static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, c memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, name); + strlcpy(id.name, name, sizeof(id.name)); id.index = index; return snd_ctl_find_id(card, &id); }

2 1

[alsa-devel] Add documentation for codec specific mixer controls of Analog codec
by Raymond Yau 04 Oct '11

04 Oct '11

Analog 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" and "6ch". According to the configuration, this also controls the jack-retasking of multi-I/O jacks. * Independent HP When this enum control is enabled, the headphone output is routed from an individual stream (the third PCM such as hw:0,2) instead of the primary stream.

2 1

[alsa-devel] [PATCH] ASoC: Remove needless codec->dapm.bias_level assignment to SND_SOC_BIAS_OFF
by Axel Lin 04 Oct '11

04 Oct '11

This assignment is done by the snd_soc_register_codec so there is no need to redo it in probe function of a codec driver. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sn95031.c | 1 - sound/soc/codecs/wm5100.c | 2 -- sound/soc/codecs/wm8996.c | 1 - 3 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index adb85bf..09c4593 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -827,7 +827,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.idle_bias_off = 1; /* PCM interface config diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index f603989..46afdf86 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2285,8 +2285,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) wm5100->codec = codec; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index b98a8f8..43e46c7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2706,7 +2706,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->fll_lock); dapm->idle_bias_off = true; - dapm->bias_level = SND_SOC_BIAS_OFF; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); if (ret != 0) { -- 1.7.4.1

3 3

[alsa-devel] Event flood on USB disconnect
by Arun Raghavan 04 Oct '11

04 Oct '11

Hi folks, We're seeing a problem when disconnecting USB headsets in PulseAudio. Steps to reproduce are basically: start PulseAudio, plug in USB device, wait for it to suspend (pcm is closed but mixer is not), disconnect. Fairly often, this results in PA getting killed because the polling thread is realtime and exceeds its rlimit. The problem is not limited to one machine or one USB headset, and I'm running kernel 3.0.4 I'm basically seeing a flood of events on the mixer pollfds (goes as high as 4+ million of them). As far as I can tell, they all contain (POLLERR|POLLNVAL). From a preliminary look at the code this seems to be expected on a card disconnect, but I assume not so often. I'm going to try to handle this by just stopping PA from listening on these fds when I get a (POLLERR|POLLNVAL), but it seems to me that this is a bug in the driver somewhere. Any ideas w.r.t. what might be going wrong or where I should be looking to debug further? Cheers, Arun

4 5

[alsa-devel] [PATCH] ASoC: Reduce the number of neigbours we mark dirty when updating power
by Mark Brown 04 Oct '11

04 Oct '11

If two widgets are not currently connected then there is no need to propagate a power state change between them as we mark the affected widgets when we change a connection. Similarly if a neighbour widget is already in the state being set for the current widget then there is no need to recheck. On one system I tested this gave: Power Path Neighbour Before: 114 1066 1327 After: 106 970 1186 which is an improvement, although relatively small. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 26 ++++++++++++++++++++------ 1 files changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 06d8cfa..e0d9971 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1215,6 +1215,21 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, + bool power, bool connect) +{ + /* If a connection is being made or broken then that update + * will have marked the peer dirty, otherwise the widgets are + * not connected and this update has no impact. */ + if (!connect) + return; + + /* If the peer is already in the state we're moving to then we + * won't have an impact on it. */ + if (power != peer->power) + dapm_mark_dirty(peer); +} + static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, struct list_head *up_list, struct list_head *down_list) @@ -1227,19 +1242,18 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, trace_snd_soc_dapm_widget_power(w, power); /* If we changed our power state perhaps our neigbours changed - * also. We're not yet smart enough to update relevant - * neighbours when we change the state of a widget, this acts - * as a proxy for that. It will notify more neighbours than - * is ideal. + * also. */ list_for_each_entry(path, &w->sources, list_sink) { if (path->source) { - dapm_mark_dirty(path->source); + dapm_widget_set_peer_power(path->source, power, + path->connect); } } list_for_each_entry(path, &w->sinks, list_source) { if (path->sink) { - dapm_mark_dirty(path->sink); + dapm_widget_set_peer_power(path->sink, power, + path->connect); } } -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC: Mark headphone, mic, speaker and line widgets as always connected
by Mark Brown 04 Oct '11

04 Oct '11

We're not actually doing any dynamic power management based on connection and output drivers (which are pretty much the same thing) are marked as unconditionally connected already. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e0d9971..214a709 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -329,7 +329,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } } break; - /* does not effect routing - always connected */ + /* does not affect routing - always connected */ case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: case snd_soc_dapm_output: @@ -341,13 +341,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - p->connect = 1; - break; - /* does effect routing - dynamically connected */ case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: case snd_soc_dapm_line: + p->connect = 1; + break; + /* does affect routing - dynamically connected */ case snd_soc_dapm_pre: case snd_soc_dapm_post: p->connect = 0; -- 1.7.6.3

1 0

Re: [alsa-devel] [PATCH resend] ALSA: firewire-speakers: fix locking
by Clemens Ladisch 04 Oct '11

04 Oct '11

Stefan Richter wrote: > On Oct 03 Takashi Iwai wrote: >> Stefan Richter wrote: >>> There is a lock inversion between fwspk->mutex and pcm->open_mutex >>> reported by lockdep when fwspk_hw_free is called. >> >> We should put cc to stable kernel, right? > > I am not sure. While I could reproduce the lockdep report reliably at > device unplugs, I never encountered a deadlock. The reported scenario requires that the same PCM device is freed twice at the same time, which is prevented by the file reference counting. Going through the entire dependency chain would require the device being unplugged, being independently close()d, and a ioctl being called with a swapped-out data buffer all happening at the same time. This doesn't look like an actual problem. Regards, Clemens

1 0

[alsa-devel] [PATCH 1/5] ASoC: Factor write of widget power out into a separate function
by Mark Brown 04 Oct '11

04 Oct '11

Split the decision about what the new power should be out from the implementation of that decision. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 29 ++++++++++++++++++----------- 1 files changed, 18 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c277228..dcbd468 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1197,6 +1197,23 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, + struct list_head *up_list, + struct list_head *down_list) +{ + if (w->power == power) + return; + + trace_snd_soc_dapm_widget_power(w, power); + + if (power) + dapm_seq_insert(w, up_list, true); + else + dapm_seq_insert(w, down_list, false); + + w->power = power; +} + static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, struct list_head *up_list, struct list_head *down_list) @@ -1241,17 +1258,7 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, } } - if (w->power == power) - break; - - trace_snd_soc_dapm_widget_power(w, power); - - if (power) - dapm_seq_insert(w, up_list, true); - else - dapm_seq_insert(w, down_list, false); - - w->power = power; + dapm_widget_set_power(w, power, up_list, down_list); break; } } -- 1.7.6.3

1 5

[alsa-devel] [PATCH] ASoC: Fix typo in 24.576MHz rate in WM5100
by Mark Brown 03 Oct '11

03 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index f88b0a3..d2f260f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1762,7 +1762,7 @@ static int wm5100_set_sysclk(struct snd_soc_codec *codec, int clk_id, fval = 1; break; case 22579200: - case 2457600: + case 24576000: fval = 2; break; default: @@ -1779,7 +1779,7 @@ static int wm5100_set_sysclk(struct snd_soc_codec *codec, int clk_id, case 6144000: case 12288000: - case 2457600: + case 24576000: audio_rate = 48000; break; -- 1.7.6.3

1 0

Re: [alsa-devel] [PATCH] ASoC: samsung: s3c-i2s-v2.c needs module.h
by Mark Brown 03 Oct '11

03 Oct '11

On Mon, Oct 03, 2011 at 12:03:19PM -0400, Paul Gortmaker wrote: > On 11-10-03 06:30 AM, Mark Brown wrote: > > That's exactly what I'm running at the minute with no problem... > Then I'm guessing you don't have the option for it enabled (as I'm pretty sure I'd have noticed if I wasn't building the driver, my systems would be a lot less noisy than it currently is. > Thanks Axel, I'll squish this into the commit that adds module.h > to all the other sound drivers that have it missing. Now that we've worked out what it's for I don't see any reason not to apply Axel's patch right now so I've done that. It would be useful if you could send your patches for module.h upstream, there's no reason to avoid sending them now and it'd be useful to cut down on the noise from small merge issues.

1 0

[alsa-devel] [PATCH 0/6] ASoC fixes from ARM randconfig builds
by Arnd Bergmann 03 Oct '11

03 Oct '11

Hi Mark, As promised, here are the remaining ASoC patches from my randconfig series. I hope I managed to get the submission right this time. I've left out the ones that you already took or NAKed, so this is shorter than I expected. Arnd Arnd Bergmann (6): ASoC: codecs/wm8682: use __devexit_p ASoC: codecs: AK4641 depends on GPIOLIB ASoC: imx: eukrea_tlv320 needs i2c ASoC: sh: use correct __iomem annotations ASoC: samsung: fix Kconfig dependencies ASoC: samsung: add missing __devexit_p() annotations sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8782.c | 2 +- sound/soc/imx/Kconfig | 1 + sound/soc/samsung/Kconfig | 4 ++++ sound/soc/samsung/ac97.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/sh/fsi.c | 10 +++++----- 7 files changed, 14 insertions(+), 9 deletions(-) -- 1.7.5.4

4 31

[alsa-devel] [PATCH] ALSA: HDA: Fix naming of input jacks for IDT parser
by David Henningsson 03 Oct '11

03 Oct '11

Actually, it's just Sigmatel that wants to make things its own way, making life harder for us in userspace. Make the driver more uniform by removing this code. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH] ALSA: HDA: Fix naming of input jacks for IDT parser
by David Henningsson 03 Oct '11

03 Oct '11

The Sigmatel/IDT parser should have the same naming convention for input jacks as the other codecs have. BugLink: http://bugs.launchpad.net/bugs/859704 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 14 +++----------- 1 files changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 987e3cf..dd977b6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4130,22 +4130,14 @@ static int stac92xx_add_jack(struct hda_codec *codec, #ifdef CONFIG_SND_HDA_INPUT_JACK int def_conf = snd_hda_codec_get_pincfg(codec, nid); int connectivity = get_defcfg_connect(def_conf); - char name[32]; - int err; if (connectivity && connectivity != AC_JACK_PORT_FIXED) return 0; - snprintf(name, sizeof(name), "%s at %s %s Jack", - snd_hda_get_jack_type(def_conf), - snd_hda_get_jack_connectivity(def_conf), - snd_hda_get_jack_location(def_conf)); - - err = snd_hda_input_jack_add(codec, nid, type, name); - if (err < 0) - return err; -#endif /* CONFIG_SND_HDA_INPUT_JACK */ + return snd_hda_input_jack_add(codec, nid, type, NULL); +#else return 0; +#endif /* CONFIG_SND_HDA_INPUT_JACK */ } static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid, -- 1.7.5.4 --------------040603080804090909080905--

1 0

[alsa-devel] [PATCH v4] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 03 Oct '11

03 Oct '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a device-specific IFACE_PCM kcontrol. Tested both with amixer and PulseAudio; with a corresponding patch passthrough modes are enabled automagically. ELD control size is set to zero in case of errors or wrong configurations. No notifications are implemented for now, it is expected that jack detection is used to reconfigure the audio outputs. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 13 +++++--- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 68 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 78 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index c34f730..f1c621d 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -318,6 +318,11 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, int size; unsigned char *buf; + /* + * ELD size is initialized to zero in caller function. If no errors and + * ELD is valid, actual eld_size is assigned in hdmi_update_eld() + */ + if (!eld->eld_valid) return -ENOENT; @@ -327,14 +332,13 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* set ELD buffer */ + buf = eld->eld_buffer; for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); @@ -356,7 +360,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, ret = hdmi_update_eld(eld, buf, size); error: - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aaefa7c..04d730f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -621,6 +621,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -645,6 +646,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3f1f6ac..3425401 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,66 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + int pin_idx; + + spec = codec->spec; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + + pin_idx = kcontrol->private_value; + uinfo->count = spec->pins[pin_idx].sink_eld.eld_size; + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + int pin_idx; + + spec = codec->spec; + pin_idx = kcontrol->private_value; + + memcpy(ucontrol->value.bytes.data, + spec->pins[pin_idx].sink_eld.eld_buffer, ELD_MAX_SIZE); + + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, + int device) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int err; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = pin_idx; + kctl->id.device = device; + + err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + if (err < 0) + return err; + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1193,6 +1253,14 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pin_idx); + + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, + pin_idx, + spec->pcm_rec[pin_idx].device); + + if (err < 0) + return err; } return 0; -- 1.7.6.2

2 1

[alsa-devel] [PATCH] ASoC: Staticise simtec_audio_resume()
by Axel Lin 03 Oct '11

03 Oct '11

It is exported via resume callback of struct dev_pm_ops rather than referenced directly and so should be staticised. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/s3c24xx_simtec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 349566f..c8d525b 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -300,7 +300,7 @@ static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) } #ifdef CONFIG_PM -int simtec_audio_resume(struct device *dev) +static int simtec_audio_resume(struct device *dev) { simtec_call_startup(pdata); return 0; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Staticise ep93xx_ac97_dai
by Axel Lin 03 Oct '11

03 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/ep93xx/ep93xx-ac97.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index c7417c7..3cd6158 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -335,7 +335,7 @@ static struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { .trigger = ep93xx_ac97_trigger, }; -struct snd_soc_dai_driver ep93xx_ac97_dai = { +static struct snd_soc_dai_driver ep93xx_ac97_dai = { .name = "ep93xx-ac97", .id = 0, .ac97_control = 1, -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: Fix setting update bits for WM8741_DACRMSB_ATTENUATION
by Axel Lin 03 Oct '11

03 Oct '11

After checking the code and datasheet, I think what we want in the second snd_soc_update_bits call is to update WM8741_DACRMSB_ATTENUATION register instead of WM8741_DACRLSB_ATTENUATION. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware handy, thus I only check the the code and datasheet. The bug is there since commit 992bee40 "ASoC: Initial WM8741 CODEC driver". I'd appreciate if someone can test this patch. Regards, Axel sound/soc/codecs/wm8741.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 78c9e5a..a42b282 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -462,7 +462,7 @@ static int wm8741_probe(struct snd_soc_codec *codec) WM8741_UPDATELM, WM8741_UPDATELM); snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: txx9: Add __exit_p at necessary place
by Axel Lin 03 Oct '11

03 Oct '11

We have __exit annotation for txx9aclc_generic_remove(), thus add __devexit_p to wrap it. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/txx9/txx9aclc-generic.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c index 6770e71..9b5e283 100644 --- a/sound/soc/txx9/txx9aclc-generic.c +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -62,7 +62,7 @@ static int __exit txx9aclc_generic_remove(struct platform_device *pdev) } static struct platform_driver txx9aclc_generic_driver = { - .remove = txx9aclc_generic_remove, + .remove = __exit_p(txx9aclc_generic_remove), .driver = { .name = "txx9aclc-generic", .owner = THIS_MODULE, -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: samsung: s3c-i2s-v2.c needs module.h
by Axel Lin 03 Oct '11

03 Oct '11

Include <linux/module.h> to fix below build error: CC sound/soc/samsung/s3c-i2s-v2.o sound/soc/samsung/s3c-i2s-v2.c:573: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:573: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:573: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c:638: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:638: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:638: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c:677: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:677: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:677: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c: In function 's3c_i2sv2_register_dai': sound/soc/samsung/s3c-i2s-v2.c:736: warning: initialization discards qualifiers from pointer target type sound/soc/samsung/s3c-i2s-v2.c: At top level: sound/soc/samsung/s3c-i2s-v2.c:754: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:754: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL' sound/soc/samsung/s3c-i2s-v2.c:754: warning: parameter names (without types) in function declaration sound/soc/samsung/s3c-i2s-v2.c:756: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/s3c-i2s-v2.c:756: warning: data definition has no type or storage class sound/soc/samsung/s3c-i2s-v2.c:756: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE' sound/soc/samsung/s3c-i2s-v2.c:756: warning: function declaration isn't a prototype make[3]: *** [sound/soc/samsung/s3c-i2s-v2.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against linux-next (20110930). sound/soc/samsung/s3c-i2s-v2.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 52074a2b..7a73380 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -16,6 +16,7 @@ * option) any later version. */ +#include <linux/module.h> #include <linux/delay.h> #include <linux/clk.h> #include <linux/io.h> -- 1.7.4.1

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Re: [alsa-devel] [PATCH resend] ALSA: firewire-speakers: fix locking
by Takashi Iwai 03 Oct '11

03 Oct '11

At Sat, 1 Oct 2011 15:57:46 +0200, Stefan Richter wrote: > > Date: Sat, 27 Aug 2011 16:45:28 +0200 > > There is a lock inversion between fwspk->mutex and pcm->open_mutex > reported by lockdep when fwspk_hw_free is called. > > Fixed by copying the fix from the same former issue in the isight > sound driver (commit f3f7c1837f6bcae3601fc535b339426868bf1549 > "ALSA: isight: fix locking"). > > Signed-off-by: Stefan Richter <stefanr(a)s5r6.in-berlin.de> We should put cc to stable kernel, right? thanks, Takashi > --- > sound/firewire/speakers.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > Index: b/sound/firewire/speakers.c > =================================================================== > --- a/sound/firewire/speakers.c > +++ b/sound/firewire/speakers.c > @@ -772,9 +772,10 @@ static int __devexit fwspk_remove(struct > { > struct fwspk *fwspk = dev_get_drvdata(dev); > > - mutex_lock(&fwspk->mutex); > amdtp_out_stream_pcm_abort(&fwspk->stream); > snd_card_disconnect(fwspk->card); > + > + mutex_lock(&fwspk->mutex); > fwspk_stop_stream(fwspk); > mutex_unlock(&fwspk->mutex); > > @@ -790,8 +791,8 @@ static void fwspk_bus_reset(struct fw_un > fcp_bus_reset(fwspk->unit); > > if (cmp_connection_update(&fwspk->connection) < 0) { > - mutex_lock(&fwspk->mutex); > amdtp_out_stream_pcm_abort(&fwspk->stream); > + mutex_lock(&fwspk->mutex); > fwspk_stop_stream(fwspk); > mutex_unlock(&fwspk->mutex); > return; > > > -- > Stefan Richter > -=====-==-== =-=- ----= > http://arcgraph.de/sr/ >

1 0

[alsa-devel] [PATCH 1/2] ASoC: Remove unused rate variable in magician_playback_hw_params
by Axel Lin 02 Oct '11

02 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/magician.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 67dcc36..098e5f4 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -92,11 +92,10 @@ static int magician_playback_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; - unsigned int acps, acds, width, rate; + unsigned int acps, acds, width; unsigned int div4 = PXA_SSP_CLK_SCDB_4; int ret = 0; - rate = params_rate(params); width = snd_pcm_format_physical_width(params_format(params)); /* -- 1.7.4.1

2 3

[alsa-devel] [PATCH 1/3] ASoC: tegra: Staticise tegra_i2s_dai and tegra_spdif_dai
by Axel Lin 02 Oct '11

02 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/tegra/tegra_i2s.c | 2 +- sound/soc/tegra/tegra_spdif.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index f36b996..6728fab 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -312,7 +312,7 @@ static struct snd_soc_dai_ops tegra_i2s_dai_ops = { .trigger = tegra_i2s_trigger, }; -struct snd_soc_dai_driver tegra_i2s_dai[] = { +static struct snd_soc_dai_driver tegra_i2s_dai[] = { { .name = DRV_NAME ".0", .probe = tegra_i2s_probe, diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index abe606b..1bd07d1 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -232,7 +232,7 @@ static struct snd_soc_dai_ops tegra_spdif_dai_ops = { .trigger = tegra_spdif_trigger, }; -struct snd_soc_dai_driver tegra_spdif_dai = { +static struct snd_soc_dai_driver tegra_spdif_dai = { .name = DRV_NAME, .probe = tegra_spdif_probe, .playback = { -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: samsung: Add __devexit_p at necessary places
by Axel Lin 02 Oct '11

02 Oct '11

According to the comments in include/linux/init.h: "Pointers to __devexit functions must use __devexit_p(function_name), the wrapper will insert either the function_name or NULL, depending on the confi options." Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Jaswinder Singh <jassi.brar(a)samsung.com> Cc: Ben Dooks <ben(a)simtec.co.uk> Cc: Seungwhan Youn <sw.youn(a)samsung.com> Cc: Jassi Brar <jassisinghbrar(a)gmail.com> --- sound/soc/samsung/ac97.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/spdif.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index bc4e234..16521e3 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -502,7 +502,7 @@ static __devexit int s3c_ac97_remove(struct platform_device *pdev) static struct platform_driver s3c_ac97_driver = { .probe = s3c_ac97_probe, - .remove = s3c_ac97_remove, + .remove = __devexit_p(s3c_ac97_remove), .driver = { .name = "samsung-ac97", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 79d4bce..bff42bf 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1137,7 +1137,7 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, - .remove = samsung_i2s_remove, + .remove = __devexit_p(samsung_i2s_remove), .driver = { .name = "samsung-i2s", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 4c191f0..05a47cf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -625,7 +625,7 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) static struct platform_driver s3c_pcm_driver = { .probe = s3c_pcm_dev_probe, - .remove = s3c_pcm_dev_remove, + .remove = __devexit_p(s3c_pcm_dev_remove), .driver = { .name = "samsung-pcm", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index f01d4c4..7bbec25 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -177,7 +177,7 @@ static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) static struct platform_driver s3c2412_iis_driver = { .probe = s3c2412_iis_dev_probe, - .remove = s3c2412_iis_dev_remove, + .remove = __devexit_p(s3c2412_iis_dev_remove), .driver = { .name = "s3c2412-iis", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index c5a30d0..558c64b 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -482,7 +482,7 @@ static __devexit int s3c24xx_iis_dev_remove(struct platform_device *pdev) static struct platform_driver s3c24xx_iis_driver = { .probe = s3c24xx_iis_dev_probe, - .remove = s3c24xx_iis_dev_remove, + .remove = __devexit_p(s3c24xx_iis_dev_remove), .driver = { .name = "s3c24xx-iis", .owner = THIS_MODULE, diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 7ca5bdc..965b2bd 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -476,7 +476,7 @@ static __devexit int spdif_remove(struct platform_device *pdev) static struct platform_driver samsung_spdif_driver = { .probe = spdif_probe, - .remove = spdif_remove, + .remove = __devexit_p(spdif_remove), .driver = { .name = "samsung-spdif", .owner = THIS_MODULE, -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: kirkwood-i2s: Add __devexit_p at necessary place
by Axel Lin 02 Oct '11

02 Oct '11

According to the comments in include/linux/init.h: "Pointers to __devexit functions must use __devexit_p(function_name), the wrapper will insert either the function_name or NULL, depending on the config options." We have __devexit annotation for kirkwood_i2s_dev_remove(), thus add __devexit_p at necessary place. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d0bcf3f..715e841 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -476,7 +476,7 @@ static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev) static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, - .remove = kirkwood_i2s_dev_remove, + .remove = __devexit_p(kirkwood_i2s_dev_remove), .driver = { .name = DRV_NAME, .owner = THIS_MODULE, -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: wm8782: Add __devexit_p at necessary place
by Axel Lin 02 Oct '11

02 Oct '11

According to the comments in include/linux/init.h: "Pointers to __devexit functions must use __devexit_p(function_name), the wrapper will insert either the function_name or NULL, depending on the config options." We have __devexit annotation for wm8782_remove(), thus add __devexit_p at necessary place. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8782.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index a2a09f8..f2ced71 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -60,7 +60,7 @@ static struct platform_driver wm8782_codec_driver = { .owner = THIS_MODULE, }, .probe = wm8782_probe, - .remove = wm8782_remove, + .remove = __devexit_p(wm8782_remove), }; static int __init wm8782_init(void) -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Fix setting update bits for WM8753_LADC and WM8753_RADC
by Axel Lin 02 Oct '11

02 Oct '11

Current code set update bits for WM8753_LDAC and WM8753_RDAC twice, but missed setting update bits for WM8753_LADC and WM8753_RADC. I think it is a copy-paste bug in commit 776065 "ASoC: codecs: wm8753: Fix register cache incoherency". Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8753.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index fe04a10..a950471 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1455,8 +1455,8 @@ static int wm8753_probe(struct snd_soc_codec *codec) /* set the update bits */ snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100); - snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100); - snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8753_LADC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8753_RADC, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_LOUT1V, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_ROUT1V, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_LOUT2V, 0x0100, 0x0100); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: omap-mcbsp: Prepare for init time DAI format setting
by Jarkko Nikula 02 Oct '11

02 Oct '11

Before commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link") expectation for omap-mcbsp was that snd_soc_dai_set_fmt is to be called first in machine hw_params callback before other CPU DAI functions. Thus it was enough that only omap_mcbsp_dai_set_dai_fmt cleared the mcbsp->regs structure. Now this doesn't hold anymore since machine drivers can set the DAI format only once on init time and thus mcbsp->regs may get out of sync when other CPU DAI functions are modifying them dynamically with different values between the calls. Therefore clear the accessed mcbsp->regs bits and bitfields in other functions too. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 894f2f3..7f70061 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -317,6 +317,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return 0; } + regs->rcr2 &= ~(RPHASE | RFRLEN2(0x7f) | RWDLEN2(7)); + regs->xcr2 &= ~(RPHASE | XFRLEN2(0x7f) | XWDLEN2(7)); + regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7)); + regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7)); format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || @@ -369,6 +373,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, framesize = wlen * channels; /* Set FS period and length in terms of bit clock periods */ + regs->srgr2 &= ~FPER(0xfff); + regs->srgr1 &= ~FWID(0xff); switch (format) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: @@ -505,6 +511,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, return -ENODEV; mcbsp_data->clk_div = div; + regs->srgr1 &= ~CLKGDV(0xff); regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -534,6 +541,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return -EINVAL; mcbsp_data->in_freq = freq; + regs->srgr2 &= ~CLKSM; + regs->pcr0 &= ~SCLKME; switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Use driver_data field of struct i2c_device_id to identify models
by Axel Lin 30 Sep '11

30 Sep '11

Save model information in driver_data so we can simplify the implementation. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 13 ++++--------- 1 files changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0963c4c..597abb9 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1495,9 +1495,9 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { */ static const struct i2c_device_id aic3x_i2c_id[] = { - [AIC3X_MODEL_3X] = { "tlv320aic3x", 0 }, - [AIC3X_MODEL_33] = { "tlv320aic33", 0 }, - [AIC3X_MODEL_3007] = { "tlv320aic3007", 0 }, + { "tlv320aic3x", AIC3X_MODEL_3X }, + { "tlv320aic33", AIC3X_MODEL_33 }, + { "tlv320aic3007", AIC3X_MODEL_3007 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1512,7 +1512,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_pdata *pdata = i2c->dev.platform_data; struct aic3x_priv *aic3x; int ret; - const struct i2c_device_id *tbl; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1531,11 +1530,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->gpio_reset = -1; } - for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) { - if (!strcmp(tbl->name, id->name)) - break; - } - aic3x->model = tbl - aic3x_i2c_id; + aic3x->model = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Add BUG() assertion if max98088_get_channel returns -EINVAL
by Axel Lin 30 Sep '11

30 Sep '11

The callers use the return value of max98088_get_channel as array index to access max98088->dai[] array. Add BUG() assertion for out of bound access of max98088->dai[] array. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/max98088.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ac65a2d..aaca91c 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1703,6 +1703,7 @@ static int max98088_get_channel(const char *name) return 0; if (strcmp(name, "EQ2 Mode") == 0) return 1; + BUG(); return -EINVAL; } -- 1.7.4.1

4 13

[alsa-devel] [PATCH v3 0/4] ASoC: twl6040: Gain ramp code cleanups
by Peter Ujfalusi 30 Sep '11

30 Sep '11

Hello, Changes since v2: - Use cancel_delayed_work_sync to make sure that we do not have pending work, or executing work when modifying the variables used by the ramp code. Intro mail from v2: Changes since v1: - Loop counters cleaned, commented in patch 2 - Use cancel_delayed_work_sync to make sure that we do not have pending work, or executing work when modifying the variables used by the ramp code. - patches already taken left out. Intro from v1: the following series cleans up the gain ramp code found in the twl6040 codec driver. Main changes: - use one workqueue for the twl6040 codec driver (instead of the original 3) - Delays between the steps does not need to be different among the range. I assume, that the original code contained copy-paste snippets from wm8350 for this part - Cleanups for the DAPM_OUT_DRV_E event handler code. - delayed_works moved within their corresponding struct. The series has been generated on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch + ASoC: omap-mcpdm/twl6040: Offset cancellation series. Side note: I have done this to better understand (while cleaning up the twl6040 driver) the requirements for the ramp code, and to study the possibility of adding support in the core for this (to handle the wm8350, and twl6040 in a generic way later). I'm still looking at the optimal implementation without ending up with too complicated code/structures... Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: One workqueue should be enough ASoC: twl6040: correct loop counters for HS/HF ramp code ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event ASoC: twl6040: Simplify code in out_drv_event for pending work check sound/soc/codecs/twl6040.c | 81 +++++++++++++++++-------------------------- 1 files changed, 32 insertions(+), 49 deletions(-) -- 1.7.6.1

2 5

[alsa-devel] [PATCH for-3.2 1/2] ASoC: omap-mcbsp: Fix FS polarity for LEFT_J, DSP_A and DSP_B formats
by Jarkko Nikula 30 Sep '11

30 Sep '11

Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link") changed DAI format flag values and we cannot simply invert anymore e.g. frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there is no anymore fixed bit position for bit-clock or frame-sync inversion. Fix this by relying only on DAI format flag values passed to us and by not making any assumption on individual bit positions. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1391ea0..894f2f3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -398,7 +398,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; + bool inv_fs = false; if (mcbsp_data->configured) return 0; @@ -430,21 +430,21 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->xcr2 |= XDATDLY(0); regs->spcr1 |= RJUST(2); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; default: /* Unsupported data format */ @@ -468,7 +468,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. @@ -489,6 +489,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + if (inv_fs == true) + regs->pcr0 ^= FSXP | FSRP; return 0; } -- 1.7.6.3

3 6

[alsa-devel] [PATCH resend] ASoC: sdp4430: Fix string for FM input name
by Peter Ujfalusi 30 Sep '11

30 Sep '11

From: "Ujfalusi, Peter" <peter.ujfalusi(a)ti.com> The name contains invalid valid character (/), which causes problems when trying to create the debugfs directory structure: ASoC: Failed to create Aux/FM Stereo In debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- Hi Mark, Liam, Could you take this patch, it has been not taken for upstream. Thanks, Peter sound/soc/omap/sdp4430.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 249d84b..79ec76d 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -88,7 +88,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_INPUT("FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -113,8 +113,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "Aux/FM Stereo In"}, - {"AFMR", NULL, "Aux/FM Stereo In"}, + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -- 1.7.6.1

2 1

[alsa-devel] [PATCH] ASoC: sn95031: Do not use static variable for channel_index
by Axel Lin 30 Sep '11

30 Sep '11

No reason to use static variable for channel_index. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sn95031.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index adb85bf..02a4857 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -103,7 +103,7 @@ static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) { int base_addr, chnl_addr; int value; - static int channel_index; + int channel_index; /* Index of the first channel in which the stop bit is set */ channel_index = find_free_channel(sn95031_codec); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Add missed regulator_unregister_notifier and regulator_bulk_free in wm8995_remove
by Axel Lin 30 Sep '11

30 Sep '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8995.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 74ae599..e05ee79 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1573,9 +1573,16 @@ static int wm8995_resume(struct snd_soc_codec *codec) static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; + int i; wm8995 = snd_soc_codec_get_drvdata(codec); wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i) + regulator_unregister_notifier(wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); + + regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); return 0; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Remove unused "control_data" field of struct cs4271_private
by Axel Lin 30 Sep '11

30 Sep '11

The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs4271.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 083aab9..23d1bd5 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -156,7 +156,6 @@ static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ enum snd_soc_control_type bus_type; - void *control_data; unsigned int mclk; bool master; bool deemph; @@ -466,8 +465,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; - codec->control_data = cs4271->control_data; - if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -555,7 +552,6 @@ static int __devinit cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->control_data = spi; cs4271->bus_type = SND_SOC_SPI; return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, @@ -595,7 +591,6 @@ static int __devinit cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->control_data = client; cs4271->bus_type = SND_SOC_I2C; return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, -- 1.7.4.1

2 3

[alsa-devel] VT1708 clicks & pops
by Forest Bond 30 Sep '11

30 Sep '11

Hi, I have two boards with a VT1708: * VIA EPIA EX15000G * VIA VB8002 I'm using VLC to play audio via PulseAudio to the analog outputs. The VB8002 sounds great, but the EX15000G has random pops and clicks in the output. Both machines are running exactly the same configuration (they are using the same pre-built OS image). I've tested on the EX15000G using a few different driver versions. One is essentially the version from Ubuntu 10.04, but rebuilt due to unrelated kernel changes. The other is alsa-driver snapshot alsa-driver-1.0.24.72.g7c48f.1314.geb1d0. You can find output from alsa-info.sh as follows: EX15000G (using Ubuntu 10.04 driver): http://www.alsa-project.org/db/?f=ea56226a7917391d6adf9078650d3a9395f51b17 EX15000G (using driver from alsa-driver-1.0.24.72.g7c48f.1314.geb1d0): http://www.alsa-project.org/db/?f=d0265991ec3e975820dc439470aea1df925ca1db VB8002 (using Ubuntu 10.04 driver): http://www.alsa-project.org/db/?f=611cc772a3fab7552bbda193b6ad7998f66a60fb Any ideas what the problem might be? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.rapidrollout.com

2 14

[alsa-devel] PATCH: [plug] Move fast_ops copy after slave configuration.
by Reilly Grant 30 Sep '11

30 Sep '11

The plug plugin copies the fast_ops structure and fast_op_arg from its slave PCM on initialization and after inserting a new plugin. This works unless the slave PCM is also an instance of the plug plugin because plug can change its fast_ops structure during _snd_pcm_hw_params. Instead of copying fast_ops in snd_pcm_plug_insert_plugins wait until the end of snd_pcm_plug_hw_params when the slave has been fully initialized. This fixes a crash when two instances of plug are instantiated back to back and the format conversion plugin inserted by the second is skipped by the first because the fast_ops pointer is not properly updated. --- src/pcm/pcm_plug.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..72456d5 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -659,8 +659,6 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, } if (err) { plug->gen.slave = new; - pcm->fast_ops = new->fast_ops; - pcm->fast_op_arg = new->fast_op_arg; } k++; } @@ -1049,6 +1047,9 @@ static int snd_pcm_plug_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); snd_pcm_unlink_appl_ptr(pcm, plug->req_slave); + + pcm->fast_ops = slave->fast_ops; + pcm->fast_op_arg = slave->fast_op_arg; snd_pcm_link_hw_ptr(pcm, slave); snd_pcm_link_appl_ptr(pcm, slave); return 0; -- 1.7.3.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: Staticise non-exported symbols in rt5631
by Mark Brown 29 Sep '11

29 Sep '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/rt5631.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 889a7dd..7e1f894 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1308,7 +1308,7 @@ static const struct pll_div codec_slave_pll_div[] = { {3072000, 12288000, 0x0a90}, }; -struct coeff_clk_div coeff_div[] = { +static struct coeff_clk_div coeff_div[] = { /* sysclk is 256fs */ {2048000, 8000 * 32, 8000, 0x1000}, {2048000, 8000 * 64, 8000, 0x0000}, @@ -1680,7 +1680,7 @@ static int rt5631_resume(struct snd_soc_codec *codec) SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5631_ops = { +static struct snd_soc_dai_ops rt5631_ops = { .hw_params = rt5631_hifi_pcm_params, .set_fmt = rt5631_hifi_codec_set_dai_fmt, .set_sysclk = rt5631_hifi_codec_set_dai_sysclk, @@ -1762,7 +1762,7 @@ static __devexit int rt5631_i2c_remove(struct i2c_client *client) return 0; } -struct i2c_driver rt5631_i2c_driver = { +static struct i2c_driver rt5631_i2c_driver = { .driver = { .name = "rt5631", .owner = THIS_MODULE, -- 1.7.6.3

1 1