February 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa wiki spam
by Eliot Blennerhassett 05 Feb '13

05 Feb '13

Hundreds of spam bot registrations dominate the recent changes: http://www.alsa-project.org/main/index.php?title=Special:RecentChanges&limi… no sign so far of spam content, but it is a bit of a worry.

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[alsa-devel] [PATCH 001/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 05 Feb '13

05 Feb '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch removes the existing max98090 driver prior to installing a more complete one. --- The original author is okay to replace his recently submitted driver with this one. "... Would you mind if we began this now by replacing the existing driver with the one attached? ..." - Jerry "... But of course I agree to replace it to your code. Please send it to upstream... " - Morimoto-san sound/soc/codecs/Kconfig | 4 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/max98090.c | 577 ------------------------------------------- 3 files changed, 0 insertions(+), 583 deletions(-) delete mode 100644 sound/soc/codecs/max98090.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a84782..0e368d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C - select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -268,9 +267,6 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate -config SND_SOC_MAX98090 - tristate - config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36..aa56312 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,7 +34,6 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o -snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -158,7 +157,6 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o -obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c deleted file mode 100644 index c9772ca..0000000 --- a/sound/soc/codecs/max98090.c +++ /dev/null @@ -1,577 +0,0 @@ -/* - * max98090.c -- MAX98090 ALSA SoC Audio driver - * based on Rev0p8 datasheet - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * Based on - * - * max98095.c - * Copyright 2011 Maxim Integrated Products - * - * https://github.com/hardkernel/linux/commit/\ - * 3417d7166b17113b3b33b0a337c74d1c7cc313df#sound/soc/codecs/max98090.c - * Copyright 2011 Maxim Integrated Products - * - * 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/i2c.h> -#include <linux/module.h> -#include <linux/regmap.h> -#include <sound/soc.h> -#include <sound/tlv.h> - -/* - * - * MAX98090 Registers Definition - * - */ - -/* RESET / STATUS / INTERRUPT REGISTERS */ -#define MAX98090_0x00_SW_RESET 0x00 -#define MAX98090_0x01_INT_STS 0x01 -#define MAX98090_0x02_JACK_STS 0x02 -#define MAX98090_0x03_INT_MASK 0x03 - -/* QUICK SETUP REGISTERS */ -#define MAX98090_0x04_SYS_CLK 0x04 -#define MAX98090_0x05_SAMPLE_RATE 0x05 -#define MAX98090_0x06_DAI_IF 0x06 -#define MAX98090_0x07_DAC_PATH 0x07 -#define MAX98090_0x08_MIC_TO_ADC 0x08 -#define MAX98090_0x09_LINE_TO_ADC 0x09 -#define MAX98090_0x0A_ANALOG_MIC_LOOP 0x0A -#define MAX98090_0x0B_ANALOG_LINE_LOOP 0x0B - -/* ANALOG INPUT CONFIGURATION REGISTERS */ -#define MAX98090_0x0D_INPUT_CONFIG 0x0D -#define MAX98090_0x0E_LINE_IN_LVL 0x0E -#define MAX98090_0x0F_LINI_IN_CFG 0x0F -#define MAX98090_0x10_MIC1_IN_LVL 0x10 -#define MAX98090_0x11_MIC2_IN_LVL 0x11 - -/* MICROPHONE CONFIGURATION REGISTERS */ -#define MAX98090_0x12_MIC_BIAS_VOL 0x12 -#define MAX98090_0x13_DIGITAL_MIC_CFG 0x13 -#define MAX98090_0x14_DIGITAL_MIC_MODE 0x14 - -/* ADC PATH AND CONFIGURATION REGISTERS */ -#define MAX98090_0x15_L_ADC_MIX 0x15 -#define MAX98090_0x16_R_ADC_MIX 0x16 -#define MAX98090_0x17_L_ADC_LVL 0x17 -#define MAX98090_0x18_R_ADC_LVL 0x18 -#define MAX98090_0x19_ADC_BIQUAD_LVL 0x19 -#define MAX98090_0x1A_ADC_SIDETONE 0x1A - -/* CLOCK CONFIGURATION REGISTERS */ -#define MAX98090_0x1B_SYS_CLK 0x1B -#define MAX98090_0x1C_CLK_MODE 0x1C -#define MAX98090_0x1D_ANY_CLK1 0x1D -#define MAX98090_0x1E_ANY_CLK2 0x1E -#define MAX98090_0x1F_ANY_CLK3 0x1F -#define MAX98090_0x20_ANY_CLK4 0x20 -#define MAX98090_0x21_MASTER_MODE 0x21 - -/* INTERFACE CONTROL REGISTERS */ -#define MAX98090_0x22_DAI_IF_FMT 0x22 -#define MAX98090_0x23_DAI_TDM_FMT1 0x23 -#define MAX98090_0x24_DAI_TDM_FMT2 0x24 -#define MAX98090_0x25_DAI_IO_CFG 0x25 -#define MAX98090_0x26_FILTER_CFG 0x26 -#define MAX98090_0x27_DAI_PLAYBACK_LVL 0x27 -#define MAX98090_0x28_EQ_PLAYBACK_LVL 0x28 - -/* HEADPHONE CONTROL REGISTERS */ -#define MAX98090_0x29_L_HP_MIX 0x29 -#define MAX98090_0x2A_R_HP_MIX 0x2A -#define MAX98090_0x2B_HP_CTR 0x2B -#define MAX98090_0x2C_L_HP_VOL 0x2C -#define MAX98090_0x2D_R_HP_VOL 0x2D - -/* SPEAKER CONFIGURATION REGISTERS */ -#define MAX98090_0x2E_L_SPK_MIX 0x2E -#define MAX98090_0x2F_R_SPK_MIX 0x2F -#define MAX98090_0x30_SPK_CTR 0x30 -#define MAX98090_0x31_L_SPK_VOL 0x31 -#define MAX98090_0x32_R_SPK_VOL 0x32 - -/* ALC CONFIGURATION REGISTERS */ -#define MAX98090_0x33_ALC_TIMING 0x33 -#define MAX98090_0x34_ALC_COMPRESSOR 0x34 -#define MAX98090_0x35_ALC_EXPANDER 0x35 -#define MAX98090_0x36_ALC_GAIN 0x36 - -/* RECEIVER AND LINE_OUTPUT REGISTERS */ -#define MAX98090_0x37_RCV_LOUT_L_MIX 0x37 -#define MAX98090_0x38_RCV_LOUT_L_CNTL 0x38 -#define MAX98090_0x39_RCV_LOUT_L_VOL 0x39 -#define MAX98090_0x3A_LOUT_R_MIX 0x3A -#define MAX98090_0x3B_LOUT_R_CNTL 0x3B -#define MAX98090_0x3C_LOUT_R_VOL 0x3C - -/* JACK DETECT AND ENABLE REGISTERS */ -#define MAX98090_0x3D_JACK_DETECT 0x3D -#define MAX98090_0x3E_IN_ENABLE 0x3E -#define MAX98090_0x3F_OUT_ENABLE 0x3F -#define MAX98090_0x40_LVL_CTR 0x40 -#define MAX98090_0x41_DSP_FILTER_ENABLE 0x41 - -/* BIAS AND POWER MODE CONFIGURATION REGISTERS */ -#define MAX98090_0x42_BIAS_CTR 0x42 -#define MAX98090_0x43_DAC_CTR 0x43 -#define MAX98090_0x44_ADC_CTR 0x44 -#define MAX98090_0x45_DEV_SHUTDOWN 0x45 - -/* REVISION ID REGISTER */ -#define MAX98090_0xFF_REV_ID 0xFF - -#define MAX98090_REG_MAX_CACHED 0x45 -#define MAX98090_REG_END 0xFF - -/* - * - * MAX98090 Registers Bit Fields - * - */ - -/* MAX98090_0x06_DAI_IF */ -#define MAX98090_DAI_IF_MASK 0x3F -#define MAX98090_RJ_M (1 << 5) -#define MAX98090_RJ_S (1 << 4) -#define MAX98090_LJ_M (1 << 3) -#define MAX98090_LJ_S (1 << 2) -#define MAX98090_I2S_M (1 << 1) -#define MAX98090_I2S_S (1 << 0) - -/* MAX98090_0x45_DEV_SHUTDOWN */ -#define MAX98090_SHDNRUN (1 << 7) - -/* codec private data */ -struct max98090_priv { - struct regmap *regmap; -}; - -static const struct reg_default max98090_reg_defaults[] = { - /* RESET / STATUS / INTERRUPT REGISTERS */ - {MAX98090_0x00_SW_RESET, 0x00}, - {MAX98090_0x01_INT_STS, 0x00}, - {MAX98090_0x02_JACK_STS, 0x00}, - {MAX98090_0x03_INT_MASK, 0x04}, - - /* QUICK SETUP REGISTERS */ - {MAX98090_0x04_SYS_CLK, 0x00}, - {MAX98090_0x05_SAMPLE_RATE, 0x00}, - {MAX98090_0x06_DAI_IF, 0x00}, - {MAX98090_0x07_DAC_PATH, 0x00}, - {MAX98090_0x08_MIC_TO_ADC, 0x00}, - {MAX98090_0x09_LINE_TO_ADC, 0x00}, - {MAX98090_0x0A_ANALOG_MIC_LOOP, 0x00}, - {MAX98090_0x0B_ANALOG_LINE_LOOP, 0x00}, - - /* ANALOG INPUT CONFIGURATION REGISTERS */ - {MAX98090_0x0D_INPUT_CONFIG, 0x00}, - {MAX98090_0x0E_LINE_IN_LVL, 0x1B}, - {MAX98090_0x0F_LINI_IN_CFG, 0x00}, - {MAX98090_0x10_MIC1_IN_LVL, 0x11}, - {MAX98090_0x11_MIC2_IN_LVL, 0x11}, - - /* MICROPHONE CONFIGURATION REGISTERS */ - {MAX98090_0x12_MIC_BIAS_VOL, 0x00}, - {MAX98090_0x13_DIGITAL_MIC_CFG, 0x00}, - {MAX98090_0x14_DIGITAL_MIC_MODE, 0x00}, - - /* ADC PATH AND CONFIGURATION REGISTERS */ - {MAX98090_0x15_L_ADC_MIX, 0x00}, - {MAX98090_0x16_R_ADC_MIX, 0x00}, - {MAX98090_0x17_L_ADC_LVL, 0x03}, - {MAX98090_0x18_R_ADC_LVL, 0x03}, - {MAX98090_0x19_ADC_BIQUAD_LVL, 0x00}, - {MAX98090_0x1A_ADC_SIDETONE, 0x00}, - - /* CLOCK CONFIGURATION REGISTERS */ - {MAX98090_0x1B_SYS_CLK, 0x00}, - {MAX98090_0x1C_CLK_MODE, 0x00}, - {MAX98090_0x1D_ANY_CLK1, 0x00}, - {MAX98090_0x1E_ANY_CLK2, 0x00}, - {MAX98090_0x1F_ANY_CLK3, 0x00}, - {MAX98090_0x20_ANY_CLK4, 0x00}, - {MAX98090_0x21_MASTER_MODE, 0x00}, - - /* INTERFACE CONTROL REGISTERS */ - {MAX98090_0x22_DAI_IF_FMT, 0x00}, - {MAX98090_0x23_DAI_TDM_FMT1, 0x00}, - {MAX98090_0x24_DAI_TDM_FMT2, 0x00}, - {MAX98090_0x25_DAI_IO_CFG, 0x00}, - {MAX98090_0x26_FILTER_CFG, 0x80}, - {MAX98090_0x27_DAI_PLAYBACK_LVL, 0x00}, - {MAX98090_0x28_EQ_PLAYBACK_LVL, 0x00}, - - /* HEADPHONE CONTROL REGISTERS */ - {MAX98090_0x29_L_HP_MIX, 0x00}, - {MAX98090_0x2A_R_HP_MIX, 0x00}, - {MAX98090_0x2B_HP_CTR, 0x00}, - {MAX98090_0x2C_L_HP_VOL, 0x1A}, - {MAX98090_0x2D_R_HP_VOL, 0x1A}, - - /* SPEAKER CONFIGURATION REGISTERS */ - {MAX98090_0x2E_L_SPK_MIX, 0x00}, - {MAX98090_0x2F_R_SPK_MIX, 0x00}, - {MAX98090_0x30_SPK_CTR, 0x00}, - {MAX98090_0x31_L_SPK_VOL, 0x2C}, - {MAX98090_0x32_R_SPK_VOL, 0x2C}, - - /* ALC CONFIGURATION REGISTERS */ - {MAX98090_0x33_ALC_TIMING, 0x00}, - {MAX98090_0x34_ALC_COMPRESSOR, 0x00}, - {MAX98090_0x35_ALC_EXPANDER, 0x00}, - {MAX98090_0x36_ALC_GAIN, 0x00}, - - /* RECEIVER AND LINE_OUTPUT REGISTERS */ - {MAX98090_0x37_RCV_LOUT_L_MIX, 0x00}, - {MAX98090_0x38_RCV_LOUT_L_CNTL, 0x00}, - {MAX98090_0x39_RCV_LOUT_L_VOL, 0x15}, - {MAX98090_0x3A_LOUT_R_MIX, 0x00}, - {MAX98090_0x3B_LOUT_R_CNTL, 0x00}, - {MAX98090_0x3C_LOUT_R_VOL, 0x15}, - - /* JACK DETECT AND ENABLE REGISTERS */ - {MAX98090_0x3D_JACK_DETECT, 0x00}, - {MAX98090_0x3E_IN_ENABLE, 0x00}, - {MAX98090_0x3F_OUT_ENABLE, 0x00}, - {MAX98090_0x40_LVL_CTR, 0x00}, - {MAX98090_0x41_DSP_FILTER_ENABLE, 0x00}, - - /* BIAS AND POWER MODE CONFIGURATION REGISTERS */ - {MAX98090_0x42_BIAS_CTR, 0x00}, - {MAX98090_0x43_DAC_CTR, 0x00}, - {MAX98090_0x44_ADC_CTR, 0x06}, - {MAX98090_0x45_DEV_SHUTDOWN, 0x00}, -}; - -static const unsigned int max98090_hp_tlv[] = { - TLV_DB_RANGE_HEAD(5), - 0x0, 0x6, TLV_DB_SCALE_ITEM(-6700, 400, 0), - 0x7, 0xE, TLV_DB_SCALE_ITEM(-4000, 300, 0), - 0xF, 0x15, TLV_DB_SCALE_ITEM(-1700, 200, 0), - 0x16, 0x1B, TLV_DB_SCALE_ITEM(-400, 100, 0), - 0x1C, 0x1F, TLV_DB_SCALE_ITEM(150, 50, 0), -}; - -static struct snd_kcontrol_new max98090_snd_controls[] = { - SOC_DOUBLE_R_TLV("Headphone Volume", MAX98090_0x2C_L_HP_VOL, - MAX98090_0x2D_R_HP_VOL, 0, 31, 0, max98090_hp_tlv), -}; - -/* Left HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x29_L_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x29_L_HP_MIX, 0, 1, 0), -}; - -/* Right HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x2A_R_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x2A_R_HP_MIX, 0, 1, 0), -}; - -static struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - /* Output */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* PGA */ - SND_SOC_DAPM_PGA("HPL Out", MAX98090_0x3F_OUT_ENABLE, 7, 0, NULL, 0), - SND_SOC_DAPM_PGA("HPR Out", MAX98090_0x3F_OUT_ENABLE, 6, 0, NULL, 0), - - /* Mixer */ - SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, - max98090_left_hp_mixer_controls, - ARRAY_SIZE(max98090_left_hp_mixer_controls)), - - SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, - max98090_right_hp_mixer_controls, - ARRAY_SIZE(max98090_right_hp_mixer_controls)), - - /* DAC */ - SND_SOC_DAPM_DAC("DACL", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 1, 0), -}; - -static struct snd_soc_dapm_route max98090_audio_map[] = { - /* Output */ - {"HPL", NULL, "HPL Out"}, - {"HPR", NULL, "HPR Out"}, - - /* PGA */ - {"HPL Out", NULL, "HPL Mixer"}, - {"HPR Out", NULL, "HPR Mixer"}, - - /* Mixer*/ - {"HPL Mixer", "DACR Switch", "DACR"}, - {"HPL Mixer", "DACL Switch", "DACL"}, - - {"HPR Mixer", "DACR Switch", "DACR"}, - {"HPR Mixer", "DACL Switch", "DACL"}, -}; - -static bool max98090_volatile(struct device *dev, unsigned int reg) -{ - if ((reg == MAX98090_0x01_INT_STS) || - (reg == MAX98090_0x02_JACK_STS) || - (reg > MAX98090_REG_MAX_CACHED)) - return true; - - return false; -} - -static int max98090_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - switch (params_rate(params)) { - case 96000: - val = 1 << 5; - break; - case 32000: - val = 1 << 4; - break; - case 48000: - val = 1 << 3; - break; - case 44100: - val = 1 << 2; - break; - case 16000: - val = 1 << 1; - break; - case 8000: - val = 1 << 0; - break; - default: - dev_err(codec->dev, "unsupported rate\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x05_SAMPLE_RATE, 0x03F, val); - - return 0; -} - -static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, 0); - - switch (freq) { - case 26000000: - val = 1 << 7; - break; - case 19200000: - val = 1 << 6; - break; - case 13000000: - val = 1 << 5; - break; - case 12288000: - val = 1 << 4; - break; - case 12000000: - val = 1 << 3; - break; - case 11289600: - val = 1 << 2; - break; - default: - dev_err(codec->dev, "Invalid master clock frequency\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x04_SYS_CLK, 0xFD, val); - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - dev_dbg(dai->dev, "sysclk is %uHz\n", freq); - - return 0; -} - -static int max98090_dai_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct snd_soc_codec *codec = dai->codec; - int is_master; - u8 val; - - /* master/slave mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - is_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - is_master = 0; - break; - default: - dev_err(codec->dev, "unsupported clock\n"); - return -EINVAL; - } - - /* format */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_I2S: - val = (is_master) ? MAX98090_I2S_M : MAX98090_I2S_S; - break; - case SND_SOC_DAIFMT_RIGHT_J: - val = (is_master) ? MAX98090_RJ_M : MAX98090_RJ_S; - break; - case SND_SOC_DAIFMT_LEFT_J: - val = (is_master) ? MAX98090_LJ_M : MAX98090_LJ_S; - break; - default: - dev_err(codec->dev, "unsupported format\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x06_DAI_IF, - MAX98090_DAI_IF_MASK, val); - - return 0; -} - -#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 -#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) - -static struct snd_soc_dai_ops max98090_dai_ops = { - .set_sysclk = max98090_dai_set_sysclk, - .set_fmt = max98090_dai_set_fmt, - .hw_params = max98090_dai_hw_params, -}; - -static struct snd_soc_dai_driver max98090_dai = { - .name = "max98090-Hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = MAX98090_RATES, - .formats = MAX98090_FORMATS, - }, - .ops = &max98090_dai_ops, -}; - -static int max98090_probe(struct snd_soc_codec *codec) -{ - struct max98090_priv *priv = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; - int ret; - - codec->control_data = priv->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - /* Device active */ - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - return 0; -} - -static int max98090_remove(struct snd_soc_codec *codec) -{ - return 0; -} - -static struct snd_soc_codec_driver soc_codec_dev_max98090 = { - .probe = max98090_probe, - .remove = max98090_remove, - .controls = max98090_snd_controls, - .num_controls = ARRAY_SIZE(max98090_snd_controls), - .dapm_widgets = max98090_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(max98090_dapm_widgets), - .dapm_routes = max98090_audio_map, - .num_dapm_routes = ARRAY_SIZE(max98090_audio_map), -}; - -static const struct regmap_config max98090_regmap = { - .reg_bits = 8, - .val_bits = 8, - .max_register = MAX98090_REG_END, - .volatile_reg = max98090_volatile, - .cache_type = REGCACHE_RBTREE, - .reg_defaults = max98090_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(max98090_reg_defaults), -}; - -static int max98090_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct max98090_priv *priv; - struct device *dev = &i2c->dev; - unsigned int val; - int ret; - - priv = devm_kzalloc(dev, sizeof(struct max98090_priv), - GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); - if (IS_ERR(priv->regmap)) { - ret = PTR_ERR(priv->regmap); - dev_err(dev, "Failed to init regmap: %d\n", ret); - return ret; - } - - i2c_set_clientdata(i2c, priv); - - ret = regmap_read(priv->regmap, MAX98090_0xFF_REV_ID, &val); - if (ret < 0) { - dev_err(dev, "Failed to read device revision: %d\n", ret); - return ret; - } - dev_info(dev, "revision 0x%02x\n", val); - - ret = snd_soc_register_codec(dev, - &soc_codec_dev_max98090, - &max98090_dai, 1); - - return ret; -} - -static int max98090_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id max98090_i2c_id[] = { - { "max98090", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); - -static struct i2c_driver max98090_i2c_driver = { - .driver = { - .name = "max98090", - .owner = THIS_MODULE, - }, - .probe = max98090_i2c_probe, - .remove = max98090_i2c_remove, - .id_table = max98090_i2c_id, -}; -module_i2c_driver(max98090_i2c_driver); - -MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); -MODULE_AUTHOR("Peter Hsiang, Kuninori Morimoto"); -MODULE_LICENSE("GPL"); -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: soc-compress: Add support for not memory mapped DSPs
by Charles Keepax 05 Feb '13

05 Feb '13

The ASoC compressed API did not implement the copy callback in its compressed ops which is required for DSPs that are not memory mapped. This patch adds a second set of compressed ops which does implement the copy callback and uses that when copy is defined in the platform compressed ops, ie. when the DSP is not memory mapped. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 36 +++++++++++++++++++++++++++++++++++- 1 files changed, 35 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea7956..f9fb112 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,6 +302,22 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int soc_compr_copy(struct snd_compr_stream *cstream, + const char __user *buf, size_t count) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + ret = platform->driver->compr_ops->copy(cstream, buf, count); + + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, @@ -315,10 +331,25 @@ static struct snd_compr_ops soc_compr_ops = { .get_codec_caps = soc_compr_get_codec_caps }; +/* ASoC Compress operations for non memory mapped DSPs */ +static struct snd_compr_ops soc_compr_nomap_ops = { + .open = soc_compr_open, + .free = soc_compr_free, + .set_params = soc_compr_set_params, + .get_params = soc_compr_get_params, + .trigger = soc_compr_trigger, + .pointer = soc_compr_pointer, + .copy = soc_compr_copy, + .ack = soc_compr_ack, + .get_caps = soc_compr_get_caps, + .get_codec_caps = soc_compr_get_codec_caps +}; + /* create a new compress */ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_compr *compr; @@ -335,7 +366,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -ENOMEM; } - compr->ops = &soc_compr_ops; + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + compr->ops = &soc_compr_nomap_ops; + else + compr->ops = &soc_compr_ops; mutex_init(&compr->lock); ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); if (ret < 0) { -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: omap-pcm: No need to set constraint at open time
by Peter Ujfalusi 05 Feb '13

05 Feb '13

The same constraint is going to be set in the snd_dmaengine_pcm_open() function, so there is no need to set it here as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-pcm.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 47bdbd4..c722c2e 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -174,23 +174,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) static int omap_pcm_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_pcm_dma_data *dma_data; - int ret; snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); - /* Ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ret = snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, - &dma_data->dma_req); - return ret; + + return snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, + &dma_data->dma_req); } static int omap_pcm_close(struct snd_pcm_substream *substream) -- 1.8.1.1

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[alsa-devel] Join my network on LinkedIn
by Barry Song via LinkedIn 05 Feb '13

05 Feb '13

LinkedIn ------------ Barry Song requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Barry Accept invitation from Barry Song http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Barry Song http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/rso/196461153/a2mu/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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[alsa-devel] [PATCH 1/2] ALSA: hda - Fix phantom jacks on VT1708
by David Henningsson 05 Feb '13

05 Feb '13

The VT1708 has no unsol event capability, and polling is set using the "Jack Detect" alsamixer control. In order not to create phantom Jack controls, temporary enable jackpoll during build_controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) Sorry, the previous version of this patch the "int err" got lost. diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9641c0e..e934c49 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -661,6 +661,18 @@ static int via_init(struct hda_codec *codec) return 0; } +static int vt1708_build_controls(struct hda_codec *codec) +{ + /* In order not to create "Phantom Jack" controls, + temporary enable jackpoll */ + int err; + int old_interval = codec->jackpoll_interval; + codec->jackpoll_interval = msecs_to_jiffies(100); + err = via_build_controls(codec); + codec->jackpoll_interval = old_interval; + return err; +} + static int vt1708_build_pcms(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -723,6 +735,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; codec->patch_ops = via_patch_ops; + codec->patch_ops.build_controls = vt1708_build_controls; codec->patch_ops.build_pcms = vt1708_build_pcms; /* clear jackpoll_interval again; it's set dynamically */ -- 1.7.9.5

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[alsa-devel] How exactly snd_pcm_hw_rule_add() should to be used?
by Peter Ujfalusi 05 Feb '13

05 Feb '13

Hi, I got confused (again) regarding to how to use correctly the snd_pcm_hw_rule_add(). Looking at the documentation [1] and to the code in kernel seams to be suggesting the opposite. [A] using the 5.6. example from [1]: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, -1); The [1] tells me that the SNDRV_PCM_HW_PARAM_FORMAT will be refined based on the SNDRV_PCM_HW_PARAM_CHANNELS. In 5.7. of [1] the reverse rule is placed: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_CHANNELS, -1); Which according to the example will refine the SNDRV_PCM_HW_PARAM_CHANNELS based on the SNDRV_PCM_HW_PARAM_FORMAT. However when looking at the code in the kernel the opposite seams to be what is actually happening. [B] using the same 5.6. example from [1]: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, -1); It looks like the SNDRV_PCM_HW_PARAM_CHANNELS should be refined based on the SNDRV_PCM_HW_PARAM_FORMAT. Which one is the correct interpretation? The reason I'm asking is that I need to have a rule which would place constraint (step) on the BUFFER_SIZE based on the used RATE, but I can not use a single snd_pcm_hw_constraint_step(,,BUFFER_TIME,) because of HW limitation. Thank you, Péter [1] http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s09.html

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[alsa-devel] [PATCH 1/2] ALSA: hda - Fix phantom jacks on VT1708
by David Henningsson 05 Feb '13

05 Feb '13

The VT1708 has no unsol event capability, and polling is set using the "Jack Detect" alsamixer control. In order not to create phantom Jack controls, temporary enable jackpoll during build_controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9641c0e..d46c448 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -661,6 +661,17 @@ static int via_init(struct hda_codec *codec) return 0; } +static int vt1708_build_controls(struct hda_codec *codec) +{ + /* In order not to create "Phantom Jack" controls, + temporary enable jackpoll */ + int old_interval = codec->jackpoll_interval; + codec->jackpoll_interval = msecs_to_jiffies(100); + err = via_build_controls(codec); + codec->jackpoll_interval = old_interval; + return err; +} + static int vt1708_build_pcms(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -723,6 +734,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; codec->patch_ops = via_patch_ops; + codec->patch_ops.build_controls = vt1708_build_controls; codec->patch_ops.build_pcms = vt1708_build_pcms; /* clear jackpoll_interval again; it's set dynamically */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm_adsp: Use asynchronous I/O to write firmware and coefficients
by Mark Brown 04 Feb '13

04 Feb '13

Allow the regmap API to use asynchronous I/O where supported to minimise the delay between transfers, reducing firmware download times. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 82 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 66 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5487a94..be45e2b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/firmware.h> +#include <linux/list.h> #include <linux/pm.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> @@ -153,6 +154,43 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +struct wm_adsp_buf { + struct list_head list; + void *buf; +}; + +static struct wm_adsp_buf *wm_adsp_buf_alloc(const void *src, size_t len, + struct list_head *list) +{ + struct wm_adsp_buf *buf = kzalloc(sizeof(*buf), GFP_KERNEL); + + if (buf == NULL) + return NULL; + + buf->buf = kmemdup(src, len, GFP_KERNEL | GFP_DMA); + if (!buf->buf) { + kfree(buf); + return NULL; + } + + if (list) + list_add_tail(&buf->list, list); + + return buf; +} + +static void wm_adsp_buf_free(struct list_head *list) +{ + while (!list_empty(list)) { + struct wm_adsp_buf *buf = list_first_entry(list, + struct wm_adsp_buf, + list); + list_del(&buf->list); + kfree(buf->buf); + kfree(buf); + } +} + #define WM_ADSP_NUM_FW 4 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { @@ -254,6 +292,7 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, static int wm_adsp_load(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); const struct firmware *firmware; struct regmap *regmap = dsp->regmap; unsigned int pos = 0; @@ -265,7 +304,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; - void *buf; + struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -420,18 +459,16 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(region->data, le32_to_cpu(region->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(region->data, + le32_to_cpu(region->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; } - ret = regmap_raw_write(regmap, reg, buf, - le32_to_cpu(region->len)); - - kfree(buf); - + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(region->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -445,12 +482,20 @@ static int wm_adsp_load(struct wm_adsp *dsp) pos += le32_to_cpu(region->len) + sizeof(*region); regions++; } - + + ret = regmap_async_complete(regmap); + if (ret != 0) { + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + goto out_fw; + } + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); out_fw: + regmap_async_complete(regmap); + wm_adsp_buf_free(&buf_list); release_firmware(firmware); out: kfree(file); @@ -655,6 +700,7 @@ out: static int wm_adsp_load_coeff(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); struct regmap *regmap = dsp->regmap; struct wmfw_coeff_hdr *hdr; struct wmfw_coeff_item *blk; @@ -664,7 +710,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; - void *buf; + struct wm_adsp_buf *buf; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -776,8 +822,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(blk->data, le32_to_cpu(blk->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(blk->data, + le32_to_cpu(blk->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; @@ -786,27 +833,30 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", file, blocks, le32_to_cpu(blk->len), reg); - ret = regmap_raw_write(regmap, reg, blk->data, - le32_to_cpu(blk->len)); + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write to %x in %s\n", file, blocks, reg, region_name); } - - kfree(buf); } pos += le32_to_cpu(blk->len) + sizeof(*blk); blocks++; } + ret = regmap_async_complete(regmap); + if (ret != 0) + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); out_fw: release_firmware(firmware); + wm_adsp_buf_free(&buf_list); out: kfree(file); return 0; -- 1.7.10.4

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[alsa-devel] [PATCH 0/4] ASoC: atmel: fixes for working as module
by Bo Shen 04 Feb '13

04 Feb '13

The following patches fixes the audio part when working as module Bo Shen (4): ASoC: atmel_pcm: make it buildable as module ASoC: atmel_ssc_dai: remove error set private data ASoC: atmel_ssc_dai: correct sequence when unload ASoC: sam9g20_wm8731: disable clock and correct sequence when unload sound/soc/atmel/atmel-pcm.h | 6 ++++-- sound/soc/atmel/atmel_ssc_dai.c | 12 +----------- sound/soc/atmel/sam9g20_wm8731.c | 6 +++--- 3 files changed, 8 insertions(+), 16 deletions(-) -- 1.7.9.5

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