September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Patch for au88x0 driver. Fix audacious issue and add auto reset in case of hardware ADB ctrl failure.
by Manuel Jander 04 Sep '09

04 Sep '09

Hi, does anybody still have a AU88x0 card ? I know they are quite ancient now, but if anyone could give the attached patch a try ? Would be nice. I made the patch against linux-2.6.30.4 http://micro.homelinux.net/~mjander/au88x0.diff.gz Best Regards, Manuel

4 4

[alsa-devel] [PATCH] ASoC: AK4671: add ak4671 codec driver
by Joonyoung Shim 04 Sep '09

04 Sep '09

The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier, Receiver-Amplifier and Headphone-Amplifier. The datasheet for the ak4671 can find at the following url: http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4671.c | 898 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 ++++++++ 4 files changed, 1060 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..a2bb659 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -96,6 +97,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_AK4671 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..13f7b4f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c new file mode 100644 index 0000000..5b8ccee --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,898 @@ +/* + * ak4671.c -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ak4671.h" + +static struct snd_soc_codec *ak4671_codec; + +/* codec private data */ +struct ak4671_priv { + struct snd_soc_codec codec; + u8 reg_cache[AK4671_CACHEREGNUM]; + + unsigned int pll_on:1; +}; + +/* ak4671 register cache & default register settings */ +static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { + 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + 0x02, /* AK4671_FORMAT_SELECT (0x03) */ + 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ + 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ + 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ + 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ + 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + 0x00, /* this register not used */ + 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ + 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ + 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ + 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ + 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ + 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ + 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ + 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ + 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ + 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ + 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ + 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ + 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ + 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ + 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ + 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ + 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ + 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ + 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +}; + +/* read ak4671 register cache */ +static inline unsigned int ak4671_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= AK4671_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* write ak4671 register cache */ +static inline void ak4671_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= AK4671_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* write to the ak4671 register space */ +static int ak4671_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4671 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + ak4671_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +/* + * LOUT1/ROUT1 output volume control: + * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB) + */ +static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1); + +/* + * LOUT2/ROUT2 output volume control: + * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB) + */ +static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1); + +/* + * LOUT3/ROUT3 output volume control: + * from -6 to 3 dB in 3 dB steps + */ +static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0); + +/* + * Mic amp gain control: + * from -15 to 30 dB in 3 dB steps + * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not + * available + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new ak4671_snd_controls[] = { + /* Common playback gain controls */ + SOC_SINGLE_TLV("Stereo Line Output1 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv), + SOC_SINGLE_TLV("Headphone Output2 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv), + SOC_SINGLE_TLV("Stereo Line Output3 Playback Volume", + AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv), + + /* Common capture gain controls */ + SOC_DOUBLE_TLV("Mic Amp Capture Volume", + AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv), +}; + +/* event handlers */ +static int ak4671_out2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg = ak4671_read_reg_cache(codec, + AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + ak4671_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = ak4671_read_reg_cache(codec, + AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + ak4671_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + } + + return 0; +} + +/* Output Mixers */ +static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +/* Input MUXs */ +static const char *ak4671_lin_mux_texts[] = + {"LIN1", "LIN2", "LIN3", "LIN4"}; +static const struct soc_enum ak4671_lin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, + ARRAY_SIZE(ak4671_lin_mux_texts), + ak4671_lin_mux_texts); +static const struct snd_kcontrol_new ak4671_lin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); + +static const char *ak4671_rin_mux_texts[] = + {"RIN1", "RIN2", "RIN3", "RIN4"}; +static const struct soc_enum ak4671_rin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, + ARRAY_SIZE(ak4671_rin_mux_texts), + ak4671_rin_mux_texts); +static const struct snd_kcontrol_new ak4671_rin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); + +static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("LIN4"), + SND_SOC_DAPM_INPUT("RIN4"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 6, 0), + SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 7, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 4, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 5, 0), + + /* PGA */ + SND_SOC_DAPM_PGA("LOUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("ROUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0, + &ak4671_lout1_mixer_controls[0], + ARRAY_SIZE(ak4671_lout1_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0, + &ak4671_rout1_mixer_controls[0], + ARRAY_SIZE(ak4671_rout1_mixer_controls)), + SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 0, 0, &ak4671_lout2_mixer_controls[0], + ARRAY_SIZE(ak4671_lout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 1, 0, &ak4671_rout2_mixer_controls[0], + ARRAY_SIZE(ak4671_rout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0, + &ak4671_lout3_mixer_controls[0], + ARRAY_SIZE(ak4671_lout3_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0, + &ak4671_rout3_mixer_controls[0], + ARRAY_SIZE(ak4671_rout3_mixer_controls)), + + /* Input MUXs */ + SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0, + &ak4671_lin_mux_control), + SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0, + &ak4671_rin_mux_control), + + /* Mic Power */ + SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Outputs */ + {"LOUT1", "NULL", "LOUT1 Mixer"}, + {"ROUT1", "NULL", "ROUT1 Mixer"}, + {"LOUT2", "NULL", "LOUT2 Mix Amp"}, + {"ROUT2", "NULL", "ROUT2 Mix Amp"}, + {"LOUT3", "NULL", "LOUT3 Mixer"}, + {"ROUT3", "NULL", "ROUT3 Mixer"}, + + {"LOUT1 Mixer", "DACL", "DAC Left"}, + {"ROUT1 Mixer", "DACR", "DAC Right"}, + {"LOUT2 Mixer", "DACHL", "DAC Left"}, + {"ROUT2 Mixer", "DACHR", "DAC Right"}, + {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT3 Mixer", "DACSL", "DAC Left"}, + {"ROUT3 Mixer", "DACSR", "DAC Right"}, + + /* Inputs */ + {"LIN MUX", "LIN1", "LIN1"}, + {"LIN MUX", "LIN2", "LIN2"}, + {"LIN MUX", "LIN3", "LIN3"}, + {"LIN MUX", "LIN4", "LIN4"}, + + {"RIN MUX", "RIN1", "RIN1"}, + {"RIN MUX", "RIN2", "RIN2"}, + {"RIN MUX", "RIN3", "RIN3"}, + {"RIN MUX", "RIN4", "RIN4"}, + + {"LIN1", NULL, "Mic Bias"}, + {"RIN1", NULL, "Mic Bias"}, + {"LIN2", NULL, "Mic Bias"}, + {"RIN2", NULL, "Mic Bias"}, + + {"ADC Left", "NULL", "LIN MUX"}, + {"ADC Right", "NULL", "RIN MUX"}, + + /* Analog Loops */ + {"LIN1 Mixing Circuit", "NULL", "LIN1"}, + {"RIN1 Mixing Circuit", "NULL", "RIN1"}, + {"LIN2 Mixing Circuit", "NULL", "LIN2"}, + {"RIN2 Mixing Circuit", "NULL", "RIN2"}, + {"LIN3 Mixing Circuit", "NULL", "LIN3"}, + {"RIN3 Mixing Circuit", "NULL", "RIN3"}, + {"LIN4 Mixing Circuit", "NULL", "LIN4"}, + {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + + {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, +}; + +static int ak4671_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4671_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; + u8 fs; + + fs = ak4671_read_reg_cache(codec, AK4671_PLL_MODE_SELECT0); + fs &= ~AK4671_FS; + + switch (params_rate(params)) { + case 8000: + fs |= AK4671_FS_8KHZ; + break; + case 12000: + fs |= AK4671_FS_12KHZ; + break; + case 16000: + fs |= AK4671_FS_16KHZ; + break; + case 24000: + fs |= AK4671_FS_24KHZ; + break; + case 11025: + fs |= AK4671_FS_11_025KHZ; + break; + case 22050: + fs |= AK4671_FS_22_05KHZ; + break; + case 32000: + fs |= AK4671_FS_32KHZ; + break; + case 44100: + fs |= AK4671_FS_44_1KHZ; + break; + case 48000: + fs |= AK4671_FS_48KHZ; + break; + default: + return -EINVAL; + } + + ak4671_write(codec, AK4671_PLL_MODE_SELECT0, fs); + + return 0; +} + +static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u8 pll; + + pll = ak4671_read_reg_cache(codec, AK4671_PLL_MODE_SELECT0); + pll &= ~AK4671_PLL; + + switch (freq) { + case 11289600: + pll |= AK4671_PLL_11_2896MHZ; + break; + case 12000000: + pll |= AK4671_PLL_12MHZ; + break; + case 12288000: + pll |= AK4671_PLL_12_288MHZ; + break; + case 13000000: + pll |= AK4671_PLL_13MHZ; + break; + case 13500000: + pll |= AK4671_PLL_13_5MHZ; + break; + case 19200000: + pll |= AK4671_PLL_19_2MHZ; + break; + case 24000000: + pll |= AK4671_PLL_24MHZ; + break; + case 26000000: + pll |= AK4671_PLL_26MHZ; + break; + case 27000000: + pll |= AK4671_PLL_27MHZ; + break; + default: + return -EINVAL; + } + + ak4671_write(codec, AK4671_PLL_MODE_SELECT0, pll); + + return 0; +} + +static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4671_priv *ak4671 = codec->private_data; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = ak4671_read_reg_cache(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ak4671->pll_on = 1; + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ak4671->pll_on = 1; + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = ak4671_read_reg_cache(codec, AK4671_FORMAT_SELECT); + format &= ~AK4671_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK4671_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK4671_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_A: + format |= AK4671_DIF_DSP_MODE; + format |= AK4671_BCKP; + format |= AK4671_MSBS; + break; + default: + return -EINVAL; + } + + /* set mode and format */ + ak4671_write(codec, AK4671_PLL_MODE_SELECT1, mode); + ak4671_write(codec, AK4671_FORMAT_SELECT, format); + + return 0; +} + +static int ak4671_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ak4671_priv *ak4671 = codec->private_data; + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (ak4671->pll_on) { + reg = ak4671_read_reg_cache(codec, + AK4671_PLL_MODE_SELECT1); + reg |= AK4671_PMPLL; + ak4671_write(codec, AK4671_PLL_MODE_SELECT1, reg); + /* pll lock time: max 40ms */ + mdelay(40); + } + + reg = ak4671_read_reg_cache(codec, + AK4671_AD_DA_POWER_MANAGEMENT); + ak4671_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + if (ak4671->pll_on) { + reg = ak4671_read_reg_cache(codec, + AK4671_PLL_MODE_SELECT1); + reg &= ~AK4671_PMPLL; + ak4671_write(codec, AK4671_PLL_MODE_SELECT1, reg); + } + + ak4671_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops ak4671_dai_ops = { + .hw_params = ak4671_hw_params, + .set_sysclk = ak4671_set_dai_sysclk, + .set_fmt = ak4671_set_dai_fmt, + + /* TODO */ +}; + +struct snd_soc_dai ak4671_dai = { + .name = "AK4671", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .ops = &ak4671_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4671_dai); + +static int ak4671_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ak4671_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4671_codec; + codec = ak4671_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ak4671_snd_controls, + ARRAY_SIZE(ak4671_snd_controls)); + ak4671_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int ak4671_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM +static int ak4671_suspend(struct platform_device *pdev, pm_message_t state) +{ + /* TODO */ + + return 0; +} + +static int ak4671_resume(struct platform_device *pdev) +{ + /* TODO */ + + return 0; +} +#else +#define ak4671_suspend NULL +#define ak4671_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, + .suspend = ak4671_suspend, + .resume = ak4671_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4671; + codec->name = "AK4671"; + codec->owner = THIS_MODULE; + codec->read = ak4671_read_reg_cache; + codec->write = ak4671_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ak4671_set_bias_level; + codec->dai = &ak4671_dai; + codec->num_dai = 1; + codec->reg_cache_size = AK4671_CACHEREGNUM; + codec->reg_cache = &ak4671->reg_cache; + + memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg)); + + ak4671_dai.dev = codec->dev; + ak4671_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; +} + +static void ak4671_unregister(struct ak4671_priv *ak4671) +{ + ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ak4671_dai); + snd_soc_unregister_codec(&ak4671->codec); + kfree(ak4671); + ak4671_codec = NULL; +} + +static int __devinit ak4671_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct ak4671_priv *ak4671; + struct snd_soc_codec *codec; + + ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + if (ak4671 == NULL) + return -ENOMEM; + + codec = &ak4671->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(client, ak4671); + codec->control_data = client; + + codec->dev = &client->dev; + + return ak4671_register(ak4671); +} + +static __devexit int ak4671_i2c_remove(struct i2c_client *client) +{ + struct ak4671_priv *ak4671 = i2c_get_clientdata(client); + + ak4671_unregister(ak4671); + + return 0; +} + +static const struct i2c_device_id ak4671_i2c_id[] = { + { "ak4671", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); + +static struct i2c_driver ak4671_i2c_driver = { + .driver = { + .name = "ak4671", + .owner = THIS_MODULE, + }, + .probe = ak4671_i2c_probe, + .remove = __devexit_p(ak4671_i2c_remove), + .id_table = ak4671_i2c_id, +}; + +static int __init ak4671_init(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_init); + +static void __exit ak4671_exit(void) +{ + i2c_del_driver(&ak4671_i2c_driver); +} +module_exit(ak4671_exit); + +MODULE_DESCRIPTION("ASoC AK4671 codec driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h new file mode 100644 index 0000000..e2fad96 --- /dev/null +++ b/sound/soc/codecs/ak4671.h @@ -0,0 +1,156 @@ +/* + * ak4671.h -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _AK4671_H +#define _AK4671_H + +#define AK4671_AD_DA_POWER_MANAGEMENT 0x00 +#define AK4671_PLL_MODE_SELECT0 0x01 +#define AK4671_PLL_MODE_SELECT1 0x02 +#define AK4671_FORMAT_SELECT 0x03 +#define AK4671_MIC_SIGNAL_SELECT 0x04 +#define AK4671_MIC_AMP_GAIN 0x05 +#define AK4671_MIXING_POWER_MANAGEMENT0 0x06 +#define AK4671_MIXING_POWER_MANAGEMENT1 0x07 +#define AK4671_OUTPUT_VOLUME_CONTROL 0x08 +#define AK4671_LOUT1_SIGNAL_SELECT 0x09 +#define AK4671_ROUT1_SIGNAL_SELECT 0x0a +#define AK4671_LOUT2_SIGNAL_SELECT 0x0b +#define AK4671_ROUT2_SIGNAL_SELECT 0x0c +#define AK4671_LOUT3_SIGNAL_SELECT 0x0d +#define AK4671_ROUT3_SIGNAL_SELECT 0x0e +#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f +#define AK4671_LOUT2_POWER_MANAGERMENT 0x10 +#define AK4671_LOUT3_POWER_MANAGERMENT 0x11 +#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12 +#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13 +#define AK4671_ALC_REFERENCE_SELECT 0x14 +#define AK4671_DIGITAL_MIXING_CONTROL 0x15 +#define AK4671_ALC_TIMER_SELECT 0x16 +#define AK4671_ALC_MODE_CONTROL 0x17 +#define AK4671_MODE_CONTROL1 0x18 +#define AK4671_MODE_CONTROL2 0x19 +#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a +#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b +#define AK4671_SIDETONE_A_CONTROL 0x1c +#define AK4671_DIGITAL_FILTER_SELECT 0x1d +#define AK4671_FIL3_COEFFICIENT0 0x1e +#define AK4671_FIL3_COEFFICIENT1 0x1f +#define AK4671_FIL3_COEFFICIENT2 0x20 +#define AK4671_FIL3_COEFFICIENT3 0x21 +#define AK4671_EQ_COEFFICIENT0 0x22 +#define AK4671_EQ_COEFFICIENT1 0x23 +#define AK4671_EQ_COEFFICIENT2 0x24 +#define AK4671_EQ_COEFFICIENT3 0x25 +#define AK4671_EQ_COEFFICIENT4 0x26 +#define AK4671_EQ_COEFFICIENT5 0x27 +#define AK4671_FIL1_COEFFICIENT0 0x28 +#define AK4671_FIL1_COEFFICIENT1 0x29 +#define AK4671_FIL1_COEFFICIENT2 0x2a +#define AK4671_FIL1_COEFFICIENT3 0x2b +#define AK4671_FIL2_COEFFICIENT0 0x2c +#define AK4671_FIL2_COEFFICIENT1 0x2d +#define AK4671_FIL2_COEFFICIENT2 0x2e +#define AK4671_FIL2_COEFFICIENT3 0x2f +#define AK4671_DIGITAL_FILTER_SELECT2 0x30 +#define AK4671_E1_COEFFICIENT0 0x32 +#define AK4671_E1_COEFFICIENT1 0x33 +#define AK4671_E1_COEFFICIENT2 0x34 +#define AK4671_E1_COEFFICIENT3 0x35 +#define AK4671_E1_COEFFICIENT4 0x36 +#define AK4671_E1_COEFFICIENT5 0x37 +#define AK4671_E2_COEFFICIENT0 0x38 +#define AK4671_E2_COEFFICIENT1 0x39 +#define AK4671_E2_COEFFICIENT2 0x3a +#define AK4671_E2_COEFFICIENT3 0x3b +#define AK4671_E2_COEFFICIENT4 0x3c +#define AK4671_E2_COEFFICIENT5 0x3d +#define AK4671_E3_COEFFICIENT0 0x3e +#define AK4671_E3_COEFFICIENT1 0x3f +#define AK4671_E3_COEFFICIENT2 0x40 +#define AK4671_E3_COEFFICIENT3 0x41 +#define AK4671_E3_COEFFICIENT4 0x42 +#define AK4671_E3_COEFFICIENT5 0x43 +#define AK4671_E4_COEFFICIENT0 0x44 +#define AK4671_E4_COEFFICIENT1 0x45 +#define AK4671_E4_COEFFICIENT2 0x46 +#define AK4671_E4_COEFFICIENT3 0x47 +#define AK4671_E4_COEFFICIENT4 0x48 +#define AK4671_E4_COEFFICIENT5 0x49 +#define AK4671_E5_COEFFICIENT0 0x4a +#define AK4671_E5_COEFFICIENT1 0x4b +#define AK4671_E5_COEFFICIENT2 0x4c +#define AK4671_E5_COEFFICIENT3 0x4d +#define AK4671_E5_COEFFICIENT4 0x4e +#define AK4671_E5_COEFFICIENT5 0x4f +#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50 +#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51 +#define AK4671_EQ_CONTRO_10KHZ 0x52 +#define AK4671_PCM_IF_CONTROL0 0x53 +#define AK4671_PCM_IF_CONTROL1 0x54 +#define AK4671_PCM_IF_CONTROL2 0x55 +#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56 +#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57 +#define AK4671_SIDETONE_VOLUME_CONTROL 0x58 +#define AK4671_DIGITAL_MIXING_CONTROL2 0x59 +#define AK4671_SAR_ADC_CONTROL 0x5a + +#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) + +/* Bitfield Definitions */ + +/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ +#define AK4671_PMVCM 0x01 + +/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */ +#define AK4671_PLL 0x0f +#define AK4671_PLL_11_2896MHZ (4 << 0) +#define AK4671_PLL_12_288MHZ (5 << 0) +#define AK4671_PLL_12MHZ (6 << 0) +#define AK4671_PLL_24MHZ (7 << 0) +#define AK4671_PLL_19_2MHZ (8 << 0) +#define AK4671_PLL_13_5MHZ (12 << 0) +#define AK4671_PLL_27MHZ (13 << 0) +#define AK4671_PLL_13MHZ (14 << 0) +#define AK4671_PLL_26MHZ (15 << 0) +#define AK4671_FS 0xf0 +#define AK4671_FS_8KHZ (0 << 4) +#define AK4671_FS_12KHZ (1 << 4) +#define AK4671_FS_16KHZ (2 << 4) +#define AK4671_FS_24KHZ (3 << 4) +#define AK4671_FS_11_025KHZ (5 << 4) +#define AK4671_FS_22_05KHZ (7 << 4) +#define AK4671_FS_32KHZ (10 << 4) +#define AK4671_FS_48KHZ (11 << 4) +#define AK4671_FS_44_1KHZ (15 << 4) + +/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */ +#define AK4671_PMPLL 0x01 +#define AK4671_M_S 0x02 + +/* AK4671_FORMAT_SELECT (0x03) Fields */ +#define AK4671_DIF 0x03 +#define AK4671_DIF_DSP_MODE (0 << 0) +#define AK4671_DIF_MSB_MODE (2 << 0) +#define AK4671_DIF_I2S_MODE (3 << 0) +#define AK4671_BCKP 0x04 +#define AK4671_MSBS 0x08 +#define AK4671_SDOD 0x10 + +/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */ +#define AK4671_MUTEN 0x04 + +extern struct snd_soc_dai ak4671_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4671; + +#endif -- 1.6.0.4

3 5

[alsa-devel] a11y: howto reduce latency with alsa's dmix plugin?
by Halim Sahin 04 Sep '09

04 Sep '09

Hello, Is there a way to reduce latency on audiocard if the dmix plugin is in use? I am asking because there is a strange problem with asistive technologies which are outputing through alsa. E. G. speech-dispatcher takes almost a half sec to stop the speech-synthesizer. If the user navigates quickly through menus, there are simultane outputs at a time. (tested with an usb logitech headset, snd-intel8x0 based card, snd-hda-intel). The same problem can be reproduced by running mplayer with alsa output driver and pressing quickly left/right arrow. You should hear several samples at a time. Using an sblive card which doesn't use dmix doesn't have such problems. Using the dmix plugin should audio much faster to avoid this problem. Please help!!! Regards Halim

2 8

Re: [alsa-devel] snd_dummy on Centos - Redhat
by Giovanni Maruzzelli 04 Sep '09

04 Sep '09

On Fri, Sep 4, 2009 at 12:03 AM, Giovanni Maruzzelli<gmaruzz.lists(a)gmail.com> wrote: > I'll test it and report back here. Still we have big problems.... 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period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2320, jiffies=117968931, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=0, jiffies=117968967, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=0, jiffies=117968967, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=64, jiffies=117968971, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=64, jiffies=117968971, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=112, jiffies=117968974, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=112, jiffies=117968974, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=0, jiffies=117969042, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2048, jiffies=117969095, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2048, jiffies=117969170, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2048, jiffies=117969170, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2192, jiffies=117969179, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2192, jiffies=117969179, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2320, jiffies=117969187, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2320, jiffies=117969187, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 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kernel: giovanni line: 306, pos=2048, jiffies=117969426, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2048, jiffies=117969426, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2176, jiffies=117969434, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2176, jiffies=117969434, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2320, jiffies=117969443, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2320, jiffies=117969443, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=0, jiffies=117969479, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=0, jiffies=117969479, buffer_size=4096, period_size=2048, rate=16000 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period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2176, jiffies=117969690, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2176, jiffies=117969690, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2320, jiffies=117969699, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=2320, jiffies=117969699, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=0, jiffies=117969735, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=0, jiffies=117969735, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=64, jiffies=117969739, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=64, jiffies=117969739, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=112, jiffies=117969742, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:51 localhost kernel: giovanni line: 306, pos=112, jiffies=117969742, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:52 localhost kernel: giovanni line: 306, pos=0, jiffies=117969810, buffer_size=4096, period_size=2048, rate=16000 Sep 4 00:35:52 localhost kernel: giovanni line: 306, pos=2048, jiffies=117969863, buffer_size=4096, period_size=2048, rate=16000

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Re: [alsa-devel] snd_dummy on Centos - Redhat
by Takashi Iwai 04 Sep '09

04 Sep '09

[Forgot to add Cc to ML, added back now] At Thu, 3 Sep 2009 21:52:54 +0200, Giovanni Maruzzelli wrote: > > I had to modify a little the patch, so maybe I ruined it. My mods are > marked with "giovanni". Ah sorry, the patch was broken. I should have compile-tested. The below is a revised one. BTW, looking through your numbers: > Sep 3 21:35:48 localhost kernel: giovanni line: 296, > dpcm->frac_buf_pos / HZ=0, jiffies=107166054, buffer_size=4096, > period_size=2048, rate=16000 > Sep 3 21:35:48 localhost kernel: giovanni line: 296, > dpcm->frac_buf_pos / HZ=48, jiffies=107166057, buffer_size=4096, > period_size=2048, rate=16000 > Sep 3 21:35:48 localhost kernel: giovanni line: 296, > dpcm->frac_buf_pos / HZ=48, jiffies=107166057, buffer_size=4096, > period_size=2048, rate=16000 > Sep 3 21:35:48 localhost kernel: giovanni line: 296, > dpcm->frac_buf_pos / HZ=80, jiffies=107166059, buffer_size=4096, > period_size=2048, rate=16000 > Sep 3 21:35:48 localhost kernel: giovanni line: 296, > dpcm->frac_buf_pos / HZ=80, jiffies=107166059, buffer_size=4096, > period_size=2048, rate=16000 > Sep 3 21:35:48 localhost kernel: giovanni line: 296, > dpcm->frac_buf_pos / HZ=0, jiffies=107166133, buffer_size=4096, > period_size=2048, rate=16000 Here, suddenly you get position 0. Then... > Sep 3 21:35:48 localhost kernel: giovanni line: 296, > dpcm->frac_buf_pos / HZ=2048, jiffies=107166182, buffer_size=4096, > period_size=2048, rate=16000 Returning to the right position again. That can be an issue. Takashi --- diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 0a798bd..f08b9c8 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -207,19 +207,18 @@ struct dummy_systimer_pcm { struct timer_list timer; unsigned long base_time; unsigned int frac_pos; /* fractional sample position (based HZ) */ + unsigned int frac_period_rest; unsigned int frac_buffer_size; /* buffer_size * HZ */ unsigned int frac_period_size; /* period_size * HZ */ unsigned int rate; + int elapsed; struct snd_pcm_substream *substream; }; static void dummy_systimer_rearm(struct dummy_systimer_pcm *dpcm) { - unsigned long frac; - - frac = dpcm->frac_pos % dpcm->frac_period_size; dpcm->timer.expires = jiffies + - (dpcm->frac_period_size + dpcm->rate - 1) / dpcm->rate; + (dpcm->frac_period_rest + dpcm->rate - 1) / dpcm->rate; add_timer(&dpcm->timer); } @@ -230,10 +229,16 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) delta = jiffies - dpcm->base_time; if (!delta) return; - dpcm->base_time = jiffies; - dpcm->frac_pos += delta * dpcm->rate; + dpcm->base_time += delta; + delta *= dpcm->rate; + dpcm->frac_pos += delta; while (dpcm->frac_pos >= dpcm->frac_buffer_size) dpcm->frac_pos -= dpcm->frac_buffer_size; + if (dpcm->frac_period_rest <= delta) { + dpcm->elapsed++; + dpcm->frac_period_rest += dpcm->frac_period_size; + } + dpcm->frac_period_rest -= delta; } static int dummy_systimer_start(struct snd_pcm_substream *substream) @@ -264,6 +269,8 @@ static int dummy_systimer_prepare(struct snd_pcm_substream *substream) dpcm->rate = runtime->rate; dpcm->frac_buffer_size = runtime->buffer_size * HZ; dpcm->frac_period_size = runtime->period_size * HZ; + dpcm->frac_period_rest = dpcm->frac_period_size; + dpcm->elapsed = 0; return 0; } @@ -272,23 +279,29 @@ static void dummy_systimer_callback(unsigned long data) { struct dummy_systimer_pcm *dpcm = (struct dummy_systimer_pcm *)data; unsigned long flags; + int elapsed = 0; spin_lock_irqsave(&dpcm->lock, flags); dummy_systimer_update(dpcm); dummy_systimer_rearm(dpcm); + elapsed = dpcm->elapsed; + dpcm->elapsed = 0; spin_unlock_irqrestore(&dpcm->lock, flags); - snd_pcm_period_elapsed(dpcm->substream); + if (elapsed) + snd_pcm_period_elapsed(dpcm->substream); } static snd_pcm_uframes_t dummy_systimer_pointer(struct snd_pcm_substream *substream) { struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; + snd_pcm_uframes_t pos; spin_lock(&dpcm->lock); dummy_systimer_update(dpcm); + pos = dpcm->frac_pos / HZ; spin_unlock(&dpcm->lock); - return dpcm->frac_pos / HZ; + return pos; } static int dummy_systimer_create(struct snd_pcm_substream *substream)

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Re: [alsa-devel] snd_dummy on Centos - Redhat
by Giovanni Maruzzelli 03 Sep '09

03 Sep '09

On Thu, Sep 3, 2009 at 7:57 PM, Takashi Iwai<tiwai(a)suse.de> wrote: > > But, *what* is actually the problem? In your post, there was no clear > description of the bug -- how is the behavior at all? Problem is: the channel driver uses Skype client, and uses an API call to make it do audio I/O to tcp sockets (to listening sockets of the channel driver itself). *BUT* the Skype client insists to have a soundcard, probably for getting its timing from. Actually interfacing with an actual soundcard is taxing badly the cpu (for dmix/dsnoop/plug overhead), takes irqs, and do not scale (you cannot have 30 Skype clients happily sharing the same soundcard). So for "heavy duty" it uses the snd-dummy. It works perfectly, and scales very well, with the Ubuntu 8.04 kernel, eg: a tickless 100HZ kernel. It works perfectly on Centos is uses the other snd drivers. It works perfectly on Centos with a compiled kernel like the Ubuntu 8.04 one. It works very bad (with glitches, accumulating delays, hiccups, etc) on Centos with the original Centos kernel (1000HZ, not tickless). The ALSA unstable snapshot, and the first patch to it you sent me do not solves the problem. Now I will try the last patches you sent me today and report here. Thanks a lot for your efforts, -giovanni

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[alsa-devel] snd_dummy on Centos - Redhat
by Giovanni Maruzzelli 03 Sep '09

03 Sep '09

Hi all, I'm the maintainer of skypiax, a channel driver for FreeSWITCH and Asterisk that allows to connect to the Skype network (eg: http://wiki.freeswitch.org/wiki/Skypiax ). I'm trying to use snd-dummy 1.0.20 as a fake soundcard on a regular Centos installation (verified with both 5.2 and 5.3, fully updated). I tested with both the packages "stable" from ATrpms.net and compiling from the source of alsa-project.org. It gives me bad timing to the program that depends on it (eg: Skype), resulting in unacceptable behavior. On the same machine, same Centos kernel and distro, if I use 1.0.20 snd-hda-intel or snd-usb, no problem. On the same machine, snd_dummy with other distro, no problem. On the same machine, same Centos, if I compile and install a kernel, snd_dummy has no problem. So, it's a specific problem of incompatibility between snd-dummy and the kernel of Centos 5.2 and 5.3 (same kernels as in RHEL5). I'm trying hard to find where the problem is, but I'm not having success. Please please please, can someone have a look at it? If more informations are needed, I'm willing to cooperate. Thanks in advance, -giovanni

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[alsa-devel] ASoC: About wm8994 codec driver
by Joonyoung Shim 03 Sep '09

03 Sep '09

Hi, Mark. I have a target including the wolfson wm8994 codec chip, so i will implement the wm8994 codec driver. I wonder if you have a plan to implement it. I don't want a duplicate work. Thank you.

2 1

[alsa-devel] <alsa-dev> RFC for OMAP4 audio driver
by hari n 03 Sep '09

03 Sep '09

Hardware Background ------------------- ABE (Audio back end) IP on the OMAP4 SoC of TI provides digital audio signal processing features (real-time mixing, equalization, etc) on the media processor. ABE allows flexible routing of the audio paths with dedicated equalizer and acoustics protection for earphone/ headphone, hands-free and vibra paths. Typically the ABE is connected to a companion analog audio chip via a dedicated McPDM (multi-channel pulse density modulation) as show at http://dev.omapzoom.org/?p=misael/kernel-audio.git;a=blob;f=sound/soc/codec… ; Proposal -------- On OMAP4, audio codec functionality is spread between ABE of OMAP4 and the companion analog audio chip. The interface between the ABE and Phoenix audio chip is proprietary McPDM interface. The proposal is to combine both ABE and Phoenix into a codec driver. OMAP4 ABE also provides other links for routing digital processed data to McBSP, McASP interfaces, which are at the same level as McPDM interface. +---------+ +---------------------+ ALSA Library --|ASOC Core|--|OMAP4 Platform Driver| +---------+ +---------------------+ | <DMA-DAI> | +-----------------------+ |ABE--<McPDM>--Phoenix--|--o Speaker/MIC +-----------------------+ ABE provides a FIFO interface (similar to McBSP FIFO) and the audio data transfer between the audio buffers and ABE FIFO is handled through the CPU DAI. Handling of the McPDM interface between the digital audio (ABE) and analog audio chip (Phoenix) is handled internally in the codec driver as part of the codec DAI operations. This RFC addresses the basic audio usecases using Phoenix companion chip with the OMAP4 audio IP block. Support and configuration for using different codecs (connectivity with BT/FM etc) using I2S/PCM interfaces are targeted for future enhancements. High Level Design ----------------- Following are the files consisting of the TI OMAP4 audio driver: sound/soc/omap/omap-pcm.c: This is the platform driver for OMAP audio. It remains the same across all OMAP chip versions. sound/soc/codecs/abe-twl6030.c: This is the codec driver interface file for OMAP4 audio. It defines the codec DAIs for the HIFI, voice and Vibra subdevices. Handles the configuration of Phoenix companion chip using i2c control interface. Handles the initialization and configuration of ABE. All codec related widget controls are also handled in this file. Both, digital (ABE) and analog (twl6030) widgets will be contained in this same driver. sound/soc/codecs/abe/*: This folder contains the ABE specific handlers. i.e reset, clock, configuration of ABE ports for different stream inputs/outputs etc. sound/soc/omap/oma-abe.c: This is the DAI driver towards the abe. sound/soc/omap/sdp4430.c: This is the machine driver for SDP4430 and defines the stream connections. Questions --------- How to handle routing of digital audio from ABE to external devices like Bluetooth/FM connectivity devices which are usually connected using McBSP interfaces?. In these scenarios, we need another DAI between the ABE (platform codec) and an external codec. ABE(platform Codec (Digital) -----> Phoenix audio codec | | +--> BT/FM codec

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Re: [alsa-devel] important screensaver
by dwmw2＠infradead.org 03 Sep '09

03 Sep '09

Please confirm the document. ++++ Attachment: No Virus found ++++ F-Secure AntiVirus - www.f-secure.com

1 0

[alsa-devel] [PATCH v3 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 02 Sep '09

02 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch enables tlv320aic3101 support on DM365 EVM and it was tested on DM365 EVM rev c. Note: this patch was created based on temp/asoc branch. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 8 ++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 21 ++++++++++++++++++++- 3 files changed, 29 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 4dfd4ad..0bf835b 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -22,6 +22,14 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci DM6446 or DM355 EVM platforms. +config SND_DM365_SOC_EVM + tristate "SoC Audio support for DaVinci DM365 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM365_EVM + select SND_DAVINCI_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..21a8e64 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -12,6 +12,7 @@ snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM365_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..d636d1a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -45,7 +45,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() + || machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -184,6 +185,14 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &evm_dai, + .num_links = 1, +}; + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -216,6 +225,13 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -249,6 +265,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; -- 1.6.0.4

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[alsa-devel] No sound with C-Media CMI8788
by Ozan Çağlayan 02 Sep '09

02 Sep '09

Hi, I have two bug reports about the $subject. One of the users had tested it with 1.0.21 and still can get no sound with it. I'm attaching the alsa-info output + relevant dmesg lines: [ 7.797723] CMI8788 0000:05:06.0: PCI INT A -> Link[APC1] -> GSI 16 (level, low) -> IRQ 16 [ 7.797794] CMI8788 0000:05:06.0: PCI INT A disabled [ 7.941267] AV200 0000:05:06.0: PCI INT A -> Link[APC1] -> GSI 16 (level, low) -> IRQ 16 [ 7.941335] AV200 0000:05:06.0: PCI INT A disabled [ 8.020782] CMI8787HiFier 0000:05:06.0: PCI INT A -> Link[APC1] -> GSI 16 (level, low) -> IRQ 16 [ 8.020849] CMI8787HiFier 0000:05:06.0: PCI INT A disabled !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Tue Sep 1 15:56:18 UTC 2009 !!Linux Distribution !!------------------ Pardus 2009 !!DMI Information !!--------------- Manufacturer: Not provided Product Name: Not provided !!Kernel Information !!------------------ Kernel release: 2.6.30.1-123 Operating System: GNU/Linux Architecture: i686 Processor: AMD Athlon(tm) 64 FX-57 Processor SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.20 Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_mpu401 !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 1 [UART ]: MPU-401 UART - MPU-401 UART MPU-401 UART at 0x330, irq 10 !!PCI Soundcards installed in the system !!-------------------------------------- 05:06.0 Multimedia audio controller: C-Media Electronics Inc CMI8788 [Oxygen HD Audio] !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 05:06.0 0401: 13f6:8788 Subsystem: 13f6:8788 !!Loaded sound module options !!-------------------------- !!Module: snd_mpu401 enable : Y,N,N,N,N,N,N,N id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -2,-2,-2,-2,-2,-2,-2,-2 irq : 10,65535,65535,65535,65535,65535,65535,65535 pnp : Y,Y,Y,Y,Y,Y,Y,Y port : 816,1,1,1,1,1,1,1 uart_enter : Y,Y,Y,Y,Y,Y,Y,Y !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 32 Sep 1 04:22 /dev/snd/controlC1 crw-rw---- 1 root audio 116, 40 Sep 1 04:22 /dev/snd/midiC1D0 crw-rw---- 1 root audio 116, 1 Sep 1 04:22 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Sep 1 04:22 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # PulseAudio plugin configuration # Let's create a virtual device "pulse" for mixer and PCM pcm.pulse { type pulse hint { description "PulseAudio Sound Server" } } ctl.pulse { type pulse hint { description "PulseAudio Sound Server" } } # Let's make it the default! pcm.!default { type pulse hint { description "Default" } } ctl.!default { type pulse hint { description "Default" } } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** ARECORD **** List of CAPTURE Hardware Devices **** !!Amixer output !!------------- !!-------Mixer controls for card 1 [UART] Card hw:1 'UART'/'MPU-401 UART at 0x330, irq 10' Mixer name : '' Components : '' Controls : 0 Simple ctrls : 0 !!Alsactl output !!------------- --startcollapse-- state.UART { control { } } --endcollapse-- !!All Loaded Modules !!------------------ Module ppp_deflate zlib_deflate ppp_async crc_ccitt ppp_generic slhc ipv6 snd_hifier snd_virtuoso snd_cs4236 snd_oxygen snd_oxygen_lib snd_wavefront snd_wss_lib nouveau ppdev snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_opl3_lib snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_mpu401 snd_mpu401_uart parport_pc drm snd_rawmidi snd_seq_device snd_timer snd pcspkr agpgart k8temp snd_page_alloc lp ns558 i2c_algo_bit soundcore gameport parport i2c_nforce2 cdc_acm asus_atk0110 brd ata_generic sata_nv pata_amd pata_acpi libata

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[alsa-devel] [PATCH v3 1/2] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 02 Sep '09

02 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Registers the platform device for McBSP on dm365. 2) Add platform data to DM365 EVM board file. 3) Set i2c address for audio codec at DM365 EVM board file. Note: This patch was created based on temp/asoc branch Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 7 ++++ arch/arm/mach-davinci/dm365.c | 45 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 56 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index f6adf79..77f815c 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -177,6 +177,8 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static struct snd_platform_data dm365_evm_snd_data; + static struct i2c_board_info i2c_info[] = { { I2C_BOARD_INFO("dm365evm_keys", 0x25), @@ -185,6 +187,9 @@ static struct i2c_board_info i2c_info[] = { I2C_BOARD_INFO("24c256", 0x50), .platform_data = &eeprom_info, }, + { + I2C_BOARD_INFO("tlv320aic3x", 0x18), + }, }; static struct davinci_i2c_platform_data i2c_pdata = { @@ -476,6 +481,8 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); + + dm365_init_asp(&dm365_evm_snd_data); } static __init void dm365_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index e815174..c8bff14 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -32,6 +32,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -456,7 +457,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("voice_codec", NULL, &voicecodec_clk), - CLK("soc-audio.0", NULL, &asp0_clk), + CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -603,6 +604,9 @@ INT_CFG(DM365, INT_IMX1_ENABLE, 24, 1, 1, false) INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) + +EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) +EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif }; @@ -806,6 +810,31 @@ static struct platform_device dm365_edma_device = { .resource = edma_resources, }; +static struct resource dm365_asp_resources[] = { + { + .start = DAVINCI_DM365_ASP0_BASE, + .end = DAVINCI_DM365_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_asp_device = { + .name = "davinci-asp", + .id = 0, + .num_resources = ARRAY_SIZE(dm365_asp_resources), + .resource = dm365_asp_resources, +}; + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -907,6 +936,20 @@ static struct davinci_soc_info davinci_soc_info_dm365 = { .sram_len = SZ_32K, }; +void __init dm365_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_MCBSP0_BDX); + davinci_cfg_reg(DM365_MCBSP0_X); + davinci_cfg_reg(DM365_MCBSP0_BFSX); + davinci_cfg_reg(DM365_MCBSP0_BDR); + davinci_cfg_reg(DM365_MCBSP0_R); + davinci_cfg_reg(DM365_MCBSP0_BFSR); + davinci_cfg_reg(DM365_EVT2_ASP_TX); + davinci_cfg_reg(DM365_EVT3_ASP_RX); + dm365_asp_device.dev.platform_data = pdata; + platform_device_register(&dm365_asp_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..fef12b9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -11,6 +11,9 @@ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 +/* Bases of dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of dm646x register banks */ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 09db434..2291c0d 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM365_EMAC_BASE (0x01D07000) #define DM365_EMAC_CNTRL_OFFSET (0x0000) @@ -25,5 +26,6 @@ #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm365_init(void); +void __init dm365_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ -- 1.6.0.4

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[alsa-devel] [PATCH v2 1/2] Davinci: DM365: Add platform device for McBSP
by miguel.aguilar＠ridgerun.com 02 Sep '09

02 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Registers the platform device for McBSP on dm365. 2) Add platform data to DM365 EVM board file. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 ++ arch/arm/mach-davinci/dm365.c | 45 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 3 ++ arch/arm/mach-davinci/include/mach/dm365.h | 2 + 4 files changed, 53 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index f6adf79..fd2db78 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -177,6 +177,8 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static struct snd_platform_data dm365_evm_snd_data; + static struct i2c_board_info i2c_info[] = { { I2C_BOARD_INFO("dm365evm_keys", 0x25), @@ -476,6 +478,8 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); + + dm365_init_asp(&dm365_evm_snd_data); } static __init void dm365_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index e815174..c8bff14 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -32,6 +32,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -456,7 +457,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("voice_codec", NULL, &voicecodec_clk), - CLK("soc-audio.0", NULL, &asp0_clk), + CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -603,6 +604,9 @@ INT_CFG(DM365, INT_IMX1_ENABLE, 24, 1, 1, false) INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) + +EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) +EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif }; @@ -806,6 +810,31 @@ static struct platform_device dm365_edma_device = { .resource = edma_resources, }; +static struct resource dm365_asp_resources[] = { + { + .start = DAVINCI_DM365_ASP0_BASE, + .end = DAVINCI_DM365_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_asp_device = { + .name = "davinci-asp", + .id = 0, + .num_resources = ARRAY_SIZE(dm365_asp_resources), + .resource = dm365_asp_resources, +}; + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -907,6 +936,20 @@ static struct davinci_soc_info davinci_soc_info_dm365 = { .sram_len = SZ_32K, }; +void __init dm365_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_MCBSP0_BDX); + davinci_cfg_reg(DM365_MCBSP0_X); + davinci_cfg_reg(DM365_MCBSP0_BFSX); + davinci_cfg_reg(DM365_MCBSP0_BDR); + davinci_cfg_reg(DM365_MCBSP0_R); + davinci_cfg_reg(DM365_MCBSP0_BFSR); + davinci_cfg_reg(DM365_EVT2_ASP_TX); + davinci_cfg_reg(DM365_EVT3_ASP_RX); + dm365_asp_device.dev.platform_data = pdata; + platform_device_register(&dm365_asp_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..fbcbed0 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + /* Bases of da850/da830 McASP0 register banks */ #define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 09db434..2291c0d 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM365_EMAC_BASE (0x01D07000) #define DM365_EMAC_CNTRL_OFFSET (0x0000) @@ -25,5 +26,6 @@ #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm365_init(void); +void __init dm365_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ -- 1.6.0.4

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[alsa-devel] Alsa on HP Pavilion dv6-1225ee audio problems
by Munzir Taha (منذر طه) 02 Sep '09

02 Sep '09

Hi sirs, I bought this http://h10010.www1.hp.com/wwpc/me/en/ho/WF06b/321957-321957-3329744-64354-6… and found the audio is not working and thankfully gnubien on #alsa pointed me to manually add snd-hda-intel: model=hp-hdx which really made the sound come out of the speakers though the laptop is not hdx series. Now when I plug the headset the Speakers are still playing (no auto sensoring?) and I had to mute them manually which could be a feature or a bug depending on how you look at it. alsa-info.sh: http://www.alsa-project.org/db/?f=dcbcc7d98ffceb95cd7491aed02064fba337bfa7 I hope this would help you properly support this nice laptop out of the box. -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#544495: mention help key
by Elimar Riesebieter 02 Sep '09

02 Sep '09

forwarded 544495 alsa-devel(a)alsa-project.org tags 544495 + wontfix thanks * jidanni(a)jidanni.org [090901 09:30 +0800] > Package: alsa-utils > Version: 1.0.20-3 > Severity: wishlist > File: /usr/bin/alsamixer > > "(Press Escape to quit)" is visible at top. > But how about there also saying what the "help" key is too please. > > Yes, one might guess "h" or "?" and indeed be lucky. > > However one might have also guessed "q" was to quit, and then need a "z" > to counteract the ill effects of "q". > > However both z and q are not mentioned on the help screen, nor is the > man page. So please mention more.... This should be addressed to the alsa developers. Distribution specific hacks are not useful in that case. Elimar -- You cannot propel yourself forward by patting yourself on the back.

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[alsa-devel] Patch for au88x0 driver. Fix audacious issue and add auto reset in case of hardware ADB ctrl failure.
by Manuel Jander 02 Sep '09

02 Sep '09

Hi, does anybody still have a AU88x0 card ? I know they are quite ancient now, but if anyone could give the attached patch a try ? Would be nice. I made the patch against linux-2.6.30.4 Best Regards, Manuel

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#544500: mention h and ? together
by Elimar Riesebieter 01 Sep '09

01 Sep '09

forwarded 544500 alsa-devel(a)alsa-project.org tags 544500 + wontfix thanks * jidanni(a)jidanni.org [090901 09:55 +0800] > Package: alsa-utils > Version: 1.0.20-3 > Severity: minor > File: /usr/share/man/man1/alsamixer.1.gz > > ? and h both give one the help screen, but are mentioned in different > places on the man page. They should be mentioned together. > > Also on the help screen, ? is mentioned, but h is not. This should be addressed to the alsa developers. Distribution specific hacks are not useful in that case. Elimar -- .~. /V\ L I N U X /( )\ >Phear the Penguin< ^^-^^

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Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 01 Sep '09

01 Sep '09

On Wednesday 05 August 2009 23:21:53 Christian Esken wrote: > > All in all, I can produce a working environment with recent 32 and 64 bit > > kernels. This is good news. :-) > > > > Below is the summary on the tested distributions. > > > > Christian > > > > > > So we got this: > > Kubuntu Karmic Alpha-2 32 Bit, 2.6.30 (works) > > Kubuntu Karmic Alpha-2 64 Bit, 2.6.30 (works) > > SuSE 11.1 64 Bit, 2.6.27 (does not work) > > SuSE 11.1 64 Bit, 2.6.27 XEN (does not work) > > SuSE 11.2 Milestone3 64Bit, 2.6.30 (test canceled, due to unrelated > > problems) => I'll see whether I can redo the SuSE11.2 tests, after doing > > a bugreport at openSuSE. OK, that last issue was solved, and I now redid the test: SuSE 11.2 Milestone6 64Bit, 2.6.31-rc6 (works) I succesfully tested all analog (5.1) channels, using: speaker-test -D surround51 -c 6 I used ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- driver-snapshot.tar.bz2 from 24.08.2009. I am just wondering, whether the new ALSA driver is it still planned for the 2.6.31 kernel?!? Christian

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Re: [alsa-devel] [alsa-announce] ALSA 1.0.21 release
by David Favor 01 Sep '09

01 Sep '09

To -> Kernel + ALSA Guru. I'm trying to understand exactly how ALSA updates work. Specifically I'm using Ubuntu Karmic (Kernel is 2.6.31-9) and I'm interested in testing all the X-Fi card driver fixes which are listed in the ALSA 1.0.21 release notes. Someone clarify the process of how ALSA and Kernel updates work. From the 1.0.20 Upgrade Guide it appears ALSA drivers are deposited in /lib/modules/... as part of the upgrade process. Can someone verify this for me. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way!

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[alsa-devel] ALC262 models
by Jure Koren 01 Sep '09

01 Sep '09

Hi! I've received a Vaio P netbook recently, which contains an ALC262 of an unknown model. Playback generally works out of the box, but I haven't yet managed to raise the capture level for the internal microphone (though it's easy to rev up the passthrough channel which makes a nice white noise without any feedback). What I'd like to know is how difficult would it be for me to actually get test out more channels than are available with any particular model= parameter setting. Some other chipsets include model=test or model=debug, but not this one. Also, is there a tool that could help me probe around the chipset to properly identify available mixer channels? If poking the code is the only way, that's ok, but I guess it'll have to wait until I have some time available. Thanks for any pointers in advance. Best regards, -- Jure Koren, unix developer

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[alsa-devel] ALSA driver - make error
by arnuld uttre 01 Sep '09

01 Sep '09

I tried to compile latest alsa drivers anf I got make error. all the relevant information is included here. I think there is a bug in source code. [root@dune alsa-driver-1.0.21]# ./configure --with-oss=yes --with-pcm-oss=yes --with-sequencer=yes --with-cards=all --with-card-options=all --prefix=/usr ................... .................... make[2]: Leaving directory `/usr/local/src/alsa/alsa-driver-1.0.21/misc' make[1]: Leaving directory `/usr/local/src/alsa/alsa-driver-1.0.21' make -C /lib/modules/2.6.18-128.4.1.el5/source SUBDIRS=/usr/local/src/alsa/alsa-driver-1.0.21 CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.18-128.4.1.el5-i686' CC [M] /usr/local/src/alsa/alsa-driver-1.0.21/acore/hwdep.o In file included from /usr/local/src/alsa/alsa-driver-1.0.21/acore/hwdep.c:1: /usr/local/src/alsa/alsa-driver-1.0.21/include/adriver.h:1786: error: redefinition of typedef ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here make[3]: *** [/usr/local/src/alsa/alsa-driver-1.0.21/acore/hwdep.o] Error 1 make[2]: *** [/usr/local/src/alsa/alsa-driver-1.0.21/acore] Error 2 make[1]: *** [_module_/usr/local/src/alsa/alsa-driver-1.0.21] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-128.4.1.el5-i686' make: *** [compile] Error 2 [root@dune alsa-driver-1.0.21]# gcc --version gcc (GCC) 4.1.2 20080704 (Red Hat 4.1.2-44) Copyright (C) 2006 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. [root@dune alsa-driver-1.0.21]# uname -a Linux dune 2.6.18-128.4.1.el5 #1 SMP Tue Aug 4 20:23:34 EDT 2009 i686 athlon i386 GNU/Linux [root@dune alsa-driver-1.0.21]# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m [root@dune alsa-driver-1.0.21]# -- http://uttre.wordpress.com/2008/05/14/the-lost-love-of-mine/

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 01 Sep '09

01 Sep '09

The following changes since commit d2c0bdaa9362c4b2ab7416420d034a0a2d1ec979: Jarkko Nikula (1): ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: new board driver to connect bfin-5xx with ad1836 codec jassi brar (1): ASoC: Debugged improper setting of PLL fields in WM8580 driver sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 128 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8580.c | 4 +- 4 files changed, 140 insertions(+), 2 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c

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[alsa-devel] Debugged improper setting of PLL fields in WM8580 driver
by jassi brar 01 Sep '09

01 Sep '09

Bug was caught while trying to use WM8580 as I2S master on SMDK. Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved by referring to WM8580A manual and setting mask value correctly and making the code to not touch 'reserved' bits of PLL4 register. Signed-off-by: Jassi <jassi.brar(a)samsung.com> diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d547347..6bded8c 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -458,12 +458,12 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; snd_soc_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0xff); + snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0x1ff); snd_soc_write(codec, WM8580_PLLA3 + offset, (pll_div.k >> 18 & 0xf) | (pll_div.n << 4)); reg = snd_soc_read(codec, WM8580_PLLA4 + offset); - reg &= ~0x3f; + reg &= ~0x1b; reg |= pll_div.prescale | pll_div.postscale << 1 | pll_div.freqmode << 3;

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[alsa-devel] [PATCH] new board driver to connect bfin-5xx with ad1836 codec
by Barry Song 01 Sep '09

01 Sep '09

Hi Mark, As discussed, the patch uses the original TDM order without rewriting. For the match between TDM slot number and audio channel number, a new API need be added. Thanks Barry Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 128 +++++++++++++++++++++++++++++++++++++ 3 files changed, 138 insertions(+), 0 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 8a4de4d..ac927ff 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -88,6 +88,14 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD1980 tristate "SoC AD1980/1 Audio support for BF5xx" depends on SND_BF5XX_AC97 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index f4d7607..87e3042 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -16,11 +16,13 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support +snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad1938-objs := bf5xx-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c new file mode 100644 index 0000000..cd361e3 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -0,0 +1,128 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 4 2009 + * Description: Board driver for ad1836 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1836.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1836; + +static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1836_ops = { + .startup = bf5xx_ad1836_startup, + .hw_params = bf5xx_ad1836_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1836_dai, + .ops = &bf5xx_ad1836_ops, +}; + +static struct snd_soc_card bf5xx_ad1836 = { + .name = "bf5xx_ad1836", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1836_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1836_snd_devdata = { + .card = &bf5xx_ad1836, + .codec_dev = &soc_codec_dev_ad1836, +}; + +static struct platform_device *bfxx_ad1836_snd_device; + +static int __init bf5xx_ad1836_init(void) +{ + int ret; + + bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1836_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata); + bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev; + ret = platform_device_add(bfxx_ad1836_snd_device); + + if (ret) + platform_device_put(bfxx_ad1836_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1836_exit(void) +{ + platform_device_unregister(bfxx_ad1836_snd_device); +} + +module_init(bf5xx_ad1836_init); +module_exit(bf5xx_ad1836_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

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